Asterisk 15.5.0-rc1 Now Available


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The Asterisk Development Team would like to announce the first
release candidate of Asterisk 15.5.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 15.5.0-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-27818 - Username bruteforce is possible when using
      ACL with PJSIP
      (Reported by John)
 * ASTERISK-27807 - iostreams: Potential DoS when client
      connection closed prematurely
      (Reported by Sean Bright)

Bugs fixed in this release:
-----------------------------------
 * 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-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-27909 - cdr: Deadlock with submit_scheduled_batch
      and submit_unscheduled_batch
      (Reported by Denis Lebedev)
 * 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-26987 - pbx_dundi: Asterisk crashes when unloading
      module pbx_dundi.so with dundi peers
      (Reported by Kirsty
      Tyerman)
 * 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-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-27848 - rtp: DTMF Breaks With telephony-event/16000

      (Reported by Dominic)
 * 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-27860 - [patch] res_pjsip: Register
      pjsip_transport_management not externally but internally.
     
      (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-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-27639 - [patch] BuildSystem: Enable IMAP storage on
      FreeBSD and DragonFly BSD.
      (Reported by Alexander Traud)
 * ASTERISK-27804 - bridge_softmix / app_confbridge: Add support
      for combining REMB reports
      (Reported by Joshua Colp)
 * ASTERISK-27418 - app_confbridge: "core show profile bridge"
      does not output "sfu" when video_mode is sfu
      (Reported by
      Carlos Chavez)
 * ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
      designator extension.
      (Reported by Alexander Traud)

Improvements made in this release:
-----------------------------------
 * 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-27793 - cppcheck identifies redundant "if"
     
      (Reported by Ilya Shipitsin)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0-rc1

Thank you for your continued support of Asterisk!

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