[asterisk-dev] Asterisk 15.6.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Aug 28 16:10:13 CDT 2018


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 15.6.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 15.6.0-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:

Bugs fixed in this release:
-----------------------------------
 * 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-28011 - chan_sip: get_refer_info() attempted unlock
      mutex 'peer' without owning it!
      (Reported by Alec Davis)
 * 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 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 Colp)
 * ASTERISK-27968 - systemd: asterisk.service
      (Reported by
      seanchann.zhou)
 * 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-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-27957 - PJSIP proposes ICE candidates on answer even
      if not in offer
      (Reported by Torrey Searle)
 * ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS
      keep-alives.
      (Reported by Alexander Traud)
 * 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)

Improvements made in this release:
-----------------------------------
 * 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 candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0-rc1

Thank you for your continued support of Asterisk!
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