[asterisk-dev] Asterisk 13.23.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Sep 5 13:06:28 CDT 2018


The Asterisk Development Team would like to announce the release of Asterisk 13.23.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.23.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
      authentification error
      (Reported by Ian Gilmour)
 * ASTERISK-28022 - res_pjsip realtime: uri column in
      ps_contacts table can be too short
      (Reported by Florian
      Floimair)
 * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock
      mutex 'peer' without owning it!
      (Reported by Alec Davis)
 * ASTERISK-28002 - When T.140 realtime text is negociated, a
      lot of debug traces are generated
      (Reported by Emmanuel
      BUU)
 * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
      LEAVEEMPTY
      (Reported by Valentin Safonov)
 * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
      offer
      (Reported by Torrey Searle)
 * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
      Do not undef s_addr.
      (Reported by Alexander Traud)
 * ASTERISK-28001 - res_pjsip_registrar: Improve performance of
      inbound handling
      (Reported by Joshua Colp)
 * ASTERISK-27999 - Wrong SRTP use status report
      (Reported
      by Salah Ahmed)
 * 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-27957 - PJSIP proposes ICE candidates on answer even
      if not in offer
      (Reported by Torrey Searle)
 * ASTERISK-27880 - [patch] pjproject_bundled: Repair
      ./configure --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-25548 - stasis: Improve message type "Use of before
      init/after destruction" error 
      (Reported by Joshua Colp)
 * ASTERISK-27972 - res_sorcery_config: Allow object name based
      matching
      (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-27938 - [patch] Compile fails with `IPTOS_MINCOST'
      undeclared.
      (Reported by Alexander Traud)
 * 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-27944 - res_pjsip_t38: Crash receiving 1xx responses
      other than 100 before 200 for T.38 reINVITE
      (Reported by
      Joshua Elson)

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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.23.0

Thank you for your continued support of Asterisk!
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