The Asterisk Development Team would like to announce the first release candidate of Asterisk 13.23.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.23.0-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:
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PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) |
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res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) |
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chan_sip: get_refer_info() attempted unlock mutex ‘peer’ without owning it! (Reported by Alec Davis) |
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When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) |
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app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) |
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rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) |
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pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) |
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res_pjsip_registrar: Improve performance of inbound handling (Reported by Joshua Colp) |
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Wrong SRTP use status report (Reported by Salah Ahmed) |
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pjsip: Race condition in 183 re transmission can result in a deadlock (Reported by Torrey Searle) |
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make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o (Reported by Majdi Bsoul) |
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[regression] menuselect compilation failure on Solaris 10 (Reported by Samuel Owens) |
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menuselect compilation failure on Solaris 10 / gcc 3.4.3 (Reported by rleasure) |
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menuselect compilation failure on Solaris 10/gcc-4.1.1 (Reported by Bob Atkins) |
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BuildSystem: Enable Jansson in Solaris 11. (Reported by Alexander Traud) |
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res_pjsip_endpoint_identifier_ip only matches against “generic string” headers (Reported by George Joseph) |
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res_rtp_asterisk: Requires OpenSSL in Developer Mode. (Reported by Alexander Traud) |
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Frack errors in stasis.c and memory leakage (Reported by Siruja Maharjan) |
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res_pjsip: Change default transport keepalive to preserve behavior (Reported by Joshua Colp) |
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PJSIP proposes ICE candidates on answer even if not in offer (Reported by Torrey Searle) |
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pjproject_bundled: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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stasis: Improve message type “Use of before init/after destruction” error (Reported by Joshua Colp) |
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res_sorcery_config: Allow object name based matching (Reported by Joshua Colp) |
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srtp: rejecting short sdes lifetimes incompatible with obihai ATAs (Reported by Nick French) |
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res_pjsip: Spurious ERROR logging when printing headers in sip_msg (Reported by Nick French) |
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pjsip modules always get -O2 even when DONT_OPTIMIZE is set (Reported by George Joseph) |
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pjproject_bundled: Disable TCP/TLS keep-alives. (Reported by Alexander Traud) |
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Compile fails with `IPTOS_MINCOST’ undeclared. (Reported by Alexander Traud) |
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res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) |
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res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers (Reported by Ross Beer) |
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res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream (Reported by Thiago Coutinho) |
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res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) (Reported by Jared Hull) |
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res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled (Reported by Torrey Searle) |
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res_pjsip: Lock inversion in transport management (Reported by Ross Beer) |
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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:
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PJSIP: Missing “party=calling”/”party=called” in Remote-Party-ID (Reported by Eric Dantie) |
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pjproject_bundled: Find shared libraries in root –with-ssl=PATH. (Reported by Alexander Traud) |
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pjsip_wizard example gives wrong info about unsupported SRV records (Reported by Jonathan Harris) |
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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-13.23.0-rc1
Thank you for your continued support of Asterisk!