<div dir="auto"><div><div class="gmail_quote"><br><br type="attribution"><blockquote class="quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>The Asterisk Development Team would like to announce the first</div><div>release candidate of Asterisk 14.5.0.</div><div>This release candidate is available for immediate download at </div><div><a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/<wbr>pub/telephony/asterisk</a></div><div><br></div><div>The release of Asterisk 14.5.0-rc1 resolves several issues reported by the</div><div>community and would have not been possible without your participation.</div><div><br></div><div>Thank you!</div><div><br></div><div>The following issues are resolved in this release candidate:</div><div><br></div><div>Bugs fixed in this release:</div><div>------------------------------<wbr>-----</div><div> * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY</div><div> events</div><div> (Reported by Ove Aursand)</div><div> * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions</div><div> could still setup the same call again.</div><div> (Reported by</div><div> Richard Mudgett)</div><div> * ASTERISK-26143 - res_rtp_asterisk: One way audio when</div><div> transcoding</div><div> (Reported by Henning Holtschneider)</div><div> * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call</div><div> leads to misleading error report</div><div> (Reported by Bob Ham)</div><div> * ASTERISK-26983 - Crash in Manager Reload when TLS Config</div><div> Changes</div><div> (Reported by Joshua Elson)</div><div> * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with</div><div> wrong eventtime</div><div> (Reported by Etienne Lessard)</div><div> * ASTERISK-26173 - func_cdr: CDR function does not permit empty</div><div> values to be assigned</div><div> (Reported by gkloepfer)</div><div> * ASTERISK-25506 - [patch]CONFBRIDGE failure after an</div><div> app_confbrige.so module reload results in segfault or</div><div> error/warning messages.</div><div> (Reported by Frederic LE FOLL)</div><div> * ASTERISK-24529 - Using AMI Action Bridge to on an already</div><div> bridged channel causes the incorrect return priority to be used</div><div><br></div><div> (Reported by Corey Farrell)</div><div> * ASTERISK-26860 - Upon RTCP reception, netsock2.c:210</div><div> ast_sockaddr_split_hostport: Port missing in (null)</div><div> </div><div> (Reported by Evers Lab)</div><div> * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address</div><div><br></div><div> (Reported by Ksenia)</div><div> * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook</div><div> </div><div> (Reported by Richard Mudgett)</div><div> * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a</div><div> res_pjsip session to be leaked.</div><div> (Reported by Richard</div><div> Mudgett)</div><div> * ASTERISK-25823 - SIGSEGV, Segmentation fault. -</div><div> ../sysdeps/x86_64/strlen.S: No such file or directory.</div><div> </div><div> (Reported by Andreas Krüger)</div><div> * ASTERISK-26926 - func_speex: Crash caused by frame with no</div><div> datalen</div><div> (Reported by Richard Kenner)</div><div> * ASTERISK-26951 - chan_sip: ACK with SDP does not update a</div><div> direct media bridge</div><div> (Reported by Jean Aunis - Prescom)</div><div> * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build</div><div> fails for non-SSE2 instrunction Linux</div><div> (Reported by</div><div> abelbeck)</div><div> * ASTERISK-26929 - pjsip: Add database tables for RLS</div><div> </div><div> (Reported by Joshua Colp)</div><div> * ASTERISK-26953 - Asterisk crash if hep.conf have some missing</div><div> parameters</div><div> (Reported by Joel Vandal)</div><div> * ASTERISK-26890 - STUN server with non-default-route transport</div><div> causes INVITE delay</div><div> (Reported by George Joseph)</div><div> * ASTERISK-26692 - res_rtp_asterisk: Crash in</div><div> dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)</div><div> </div><div> (Reported by scgm11)</div><div> * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP</div><div> address string</div><div> (Reported by Niklas Larsson)</div><div> * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when</div><div> receiving packet</div><div> (Reported by Adagio)</div><div> * ASTERISK-26613 - format_wav: wav16 format read file only by</div><div> 320 - half of frame</div><div> (Reported by Vitaly K)</div><div> * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in</div><div> MixMonitor</div><div> (Reported by Ivan Myalkin)</div><div> * ASTERISK-21856 - STUN never works when asterisk started</div><div> without internet access</div><div> (Reported by Jeremy Kister)</div><div> * ASTERISK-20984 - Audible clicks when playing sox encoded au</div><div> file with STREAM FILE AGI command</div><div> (Reported by Roman S.)</div><div> * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in</div><div> ast_str_case_hash</div><div> (Reported by Badalian Vyacheslav)</div><div> * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use</div><div> same IP as explicit transport</div><div> (Reported by Richard Begg)</div><div> * ASTERISK-26903 - Listening TCP/TLS sockets stop when</div><div> temporarily out of open files</div><div> (Reported by Walter Doekes)</div><div> * ASTERISK-26928 - pjsip: Add database tables for PUBLISH</div><div> support</div><div> (Reported by Joshua Colp)</div><div> * ASTERISK-26927 - pjproject_bundled: Crash on</div><div> pj_ssl_get_info() while ioqueue_on_read_complete().</div><div> </div><div> (Reported by Alexander Traud)</div><div> * ASTERISK-26905 - pjproject_bundled: Merge 3 upstream</div><div> deadlock patches into bundled</div><div> (Reported by Ross Beer)</div><div> * ASTERISK-26897 - chan_sip: Security vulnerability with client</div><div> code header</div><div> (Reported by Alex Villacís Lasso)</div><div> * ASTERISK-25974 - Unused realtime MOH classes not purged on</div><div> 'moh reload'</div><div> (Reported by Sébastien Couture)</div><div> * ASTERISK-26916 - res_pjsip: Excessive refcount reached on</div><div> transport ao2 object</div><div> (Reported by Ross Beer)</div><div> * ASTERISK-21721 - SIP Failed to parse multiple Supported:</div><div> headers</div><div> (Reported by Olle Johansson)</div><div> * ASTERISK-26915 - chan_sip: Session Timers required but</div><div> refused wrongly.</div><div> (Reported by Alexander Traud)</div><div> * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not</div><div> authenticated even after receiving a 407 error code</div><div> </div><div> (Reported by Yaacov Akiba Slama)</div><div> * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn</div><div> with large app_args causes ABRT</div><div> (Reported by twisted)</div><div> * ASTERISK-26705 - libasteriskssl.so not found when asterisk is</div><div> installed for the 1st time</div><div> (Reported by George Joseph)</div><div> * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ</div><div> when creating pubsub unsubscription on client</div><div> (Reported by</div><div> Marcello Ceschia)</div><div> * ASTERISK-25490 - [patch]SDP crypto tag is validated</div><div> incorrectly</div><div> (Reported by Joerg Sonnenberger)</div><div> * ASTERISK-26086 - res_musiconhold: format option is not</div><div> documented adequately</div><div> (Reported by Jens Bürger)</div><div> * ASTERISK-23996 - No core dumps because of res_musiconhold</div><div> chdir.</div><div> (Reported by Walter Doekes)</div><div> * ASTERISK-24712 - xmpp: starttls problem causes connection</div><div> spew</div><div> (Reported by Matthias Urlichs)</div><div> * ASTERISK-26814 - pjproject_bundled build fails to download</div><div> pjproject source when using cURL</div><div> (Reported by Gergely</div><div> Dömsödi)</div><div> * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for</div><div> unavailable clients</div><div> (Reported by Anthony Critelli)</div><div> * ASTERISK-21855 - Asterisk crashes when XMPP message is sent</div><div> (JabberSend) and no internet connection is available</div><div> </div><div> (Reported by Jeremy Kister)</div><div> * ASTERISK-25622 - WARNING for "JABBER: socket read error"</div><div> should be more specific</div><div> (Reported by Sean Darcy)</div><div> * ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a</div><div> per-session basis</div><div> (Reported by Joshua Colp)</div><div> * ASTERISK-26818 - cdr: Problem setting variables in h exten</div><div> </div><div> (Reported by scgm11)</div><div> * ASTERISK-26875 - app_mixmonitor: Recording out of sync when</div><div> 183 but no RTP</div><div> (Reported by Aaron An)</div><div><br></div><div>Improvements made in this release:</div><div>------------------------------<wbr>-----</div><div> * ASTERISK-26088 - Investigate heavy memory utilization by</div><div> res_pjsip_pubsub</div><div> (Reported by Richard Mudgett)</div><div> * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report</div><div> channel name with res_hep_rtcp when using chan_sip</div><div> </div><div> (Reported by Nir Simionovich (GreenfieldTech - Israel))</div><div><br></div><div>For a full list of changes in this release candidate, please see the ChangeLog:</div><div><a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0-rc1" target="_blank">http://downloads.asterisk.org/<wbr>pub/telephony/asterisk/<wbr>ChangeLog-14.5.0-rc1</a></div><div><br></div><div>Thank you for your continued support of Asterisk!</div></div>
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