Asterisk 13.23.0-rc1 Now Available

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
https://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:
———————————–

 

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

 

PJSIP: Missing “party=calling”/”party=called” in Remote-Party-ID
(Reported by Eric Dantie)
pjproject_bundled: Find shared libraries in root –with-ssl=PATH.
(Reported by Alexander Traud)
pjsip_wizard example gives wrong info about unsupported SRV records
(Reported by Jonathan Harris)
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:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.23.0-rc1

Thank you for your continued support of Asterisk!

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