The Asterisk Development Team would like to announce the first release candidate of Asterisk 14.5.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.5.0-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:
Bugs fixed in this release:
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Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) |
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res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) |
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res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) |
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tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) |
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Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) |
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cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) |
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func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) |
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CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) |
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Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) |
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Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) |
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chan_sip: tcpbind uses wrong source address (Reported by Ksenia) |
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res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) |
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res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) |
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SIGSEGV, Segmentation fault. – ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) |
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func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) |
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chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis – Prescom) |
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pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) |
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pjsip: Add database tables for RLS (Reported by Joshua Colp) |
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Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) |
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STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) |
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res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) |
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res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) |
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res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) |
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format_wav: wav16 format read file only by 320 – half of frame (Reported by Vitaly K) |
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format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) |
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STUN never works when asterisk started without internet access (Reported by Jeremy Kister) |
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Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) |
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[UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) |
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res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) |
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Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) |
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pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) |
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pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) |
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pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) |
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chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) |
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Unused realtime MOH classes not purged on ‘moh reload’ (Reported by Sébastien Couture) |
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res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) |
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SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) |
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chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) |
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res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) |
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Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) |
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libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) |
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xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) |
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SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) |
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res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) |
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No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) |
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xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) |
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pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) |
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JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) |
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Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available (Reported by Jeremy Kister) |
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WARNING for “JABBER: socket read error” should be more specific (Reported by Sean Darcy) |
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rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) |
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cdr: Problem setting variables in h exten (Reported by scgm11) |
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app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) |
Improvements made in this release:
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Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) |
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res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech – Israel)) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0-rc1
Thank you for your continued support of Asterisk!