<div dir="ltr"><div class="gmail_quote"><div>The Asterisk Development Team would like to announce the release of Asterisk 14.5.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a>
<p>
The release of Asterisk 14.5.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
</p><p>
<b>Thank you!</b><br>
</p><p>
The following issues are resolved in this release:<br>
</p><p>
<b>Bugs fixed in this release:</b><br>
-----------------------------------<br>
</p><table border="0">
<tbody><tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982" target="_blank">ASTERISK-26982</a>] - </li></ul></td><td></td><td>chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br>(Reported by Stefan Engström)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979" target="_blank">ASTERISK-26979</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br>(Reported by Javier Riveros )</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25665" target="_blank">ASTERISK-25665</a>] - </li></ul></td><td></td><td>Duplicate logging in queue log for EXITEMPTY events<br>(Reported by Ove Aursand)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26998" target="_blank">ASTERISK-26998</a>] - </li></ul></td><td></td><td>res_pjsip_session: INVITE retransmissions could still setup the same call again.<br>(Reported by Richard Mudgett)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26143" target="_blank">ASTERISK-26143</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: One way audio when transcoding<br>(Reported by Henning Holtschneider)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26606" target="_blank">ASTERISK-26606</a>] - </li></ul></td><td></td><td>tcptls: Incorrect OpenSSL function call leads to misleading error report<br>(Reported by Bob Ham)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26983" target="_blank">ASTERISK-26983</a>] - </li></ul></td><td></td><td>Crash in Manager Reload when TLS Config Changes<br>(Reported by Joshua Elson)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25032" target="_blank">ASTERISK-25032</a>] - </li></ul></td><td></td><td>[patch]cel_odbc sometimes inserts CEL with wrong eventtime<br>(Reported by Etienne Lessard)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26173" target="_blank">ASTERISK-26173</a>] - </li></ul></td><td></td><td>func_cdr: CDR function does not permit empty values to be assigned<br>(Reported by gkloepfer)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25506" target="_blank">ASTERISK-25506</a>] - </li></ul></td><td></td><td>[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.<br>(Reported by Frederic LE FOLL)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24529" target="_blank">ASTERISK-24529</a>] - </li></ul></td><td></td><td>Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used<br>(Reported by Corey Farrell)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26860" target="_blank">ASTERISK-26860</a>] - </li></ul></td><td></td><td>Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)<br>(Reported by Evers Lab)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26922" target="_blank">ASTERISK-26922</a>] - </li></ul></td><td></td><td>chan_sip: tcpbind uses wrong source address<br>(Reported by Ksenia)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26974" target="_blank">ASTERISK-26974</a>] - </li></ul></td><td></td><td>res_pjsip: Deadlock in T.38 framehook<br>(Reported by Richard Mudgett)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26908" target="_blank">ASTERISK-26908</a>] - </li></ul></td><td></td><td>res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.<br>(Reported by Richard Mudgett)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25823" target="_blank">ASTERISK-25823</a>] - </li></ul></td><td></td><td>SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.<br>(Reported by Andreas Krüger)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26926" target="_blank">ASTERISK-26926</a>] - </li></ul></td><td></td><td>func_speex: Crash caused by frame with no datalen<br>(Reported by Richard Kenner)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26951" target="_blank">ASTERISK-26951</a>] - </li></ul></td><td></td><td>chan_sip: ACK with SDP does not update a direct media bridge<br>(Reported by Jean Aunis - Prescom)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26930" target="_blank">ASTERISK-26930</a>] - </li></ul></td><td></td><td>pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux<br>(Reported by abelbeck)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26929" target="_blank">ASTERISK-26929</a>] - </li></ul></td><td></td><td>pjsip: Add database tables for RLS<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26953" target="_blank">ASTERISK-26953</a>] - </li></ul></td><td></td><td>Asterisk crash if hep.conf have some missing parameters<br>(Reported by Joel Vandal)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26890" target="_blank">ASTERISK-26890</a>] - </li></ul></td><td></td><td>STUN server with non-default-route transport causes INVITE delay<br>(Reported by George Joseph)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26692" target="_blank">ASTERISK-26692</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)<br>(Reported by scgm11)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26835" target="_blank">ASTERISK-26835</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: Crash when freeing RTCP address string<br>(Reported by Niklas Larsson)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26853" target="_blank">ASTERISK-26853</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: Crash in pjnath when receiving packet<br>(Reported by Adagio)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26613" target="_blank">ASTERISK-26613</a>] - </li></ul></td><td></td><td>format_wav: wav16 format read file only by 320 - half of frame<br>(Reported by Vitaly K)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26169" target="_blank">ASTERISK-26169</a>] - </li></ul></td><td></td><td>format_ogg_vorbis: Memory leak using OGG in MixMonitor<br>(Reported by Ivan Myalkin)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21856" target="_blank">ASTERISK-21856</a>] - </li></ul></td><td></td><td>STUN never works when asterisk started without internet access<br>(Reported by Jeremy Kister)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20984" target="_blank">ASTERISK-20984</a>] - </li></ul></td><td></td><td>Audible clicks when playing sox encoded au file with STREAM FILE AGI command<br>(Reported by Roman S.)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26528" target="_blank">ASTERISK-26528</a>] - </li></ul></td><td></td><td>[UBSAN] strings.h:signed integer overflow in ast_str_case_hash<br>(Reported by Badalian Vyacheslav)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26851" target="_blank">ASTERISK-26851</a>] - </li></ul></td><td></td><td>res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport<br>(Reported by Richard Begg)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26903" target="_blank">ASTERISK-26903</a>] - </li></ul></td><td></td><td>Listening TCP/TLS sockets stop when temporarily out of open files<br>(Reported by Walter Doekes)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26928" target="_blank">ASTERISK-26928</a>] - </li></ul></td><td></td><td>pjsip: Add database tables for PUBLISH support<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26927" target="_blank">ASTERISK-26927</a>] - </li></ul></td><td></td><td>pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().<br>(Reported by Alexander Traud)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26905" target="_blank">ASTERISK-26905</a>] - </li></ul></td><td></td><td>pjproject_bundled: Merge 3 upstream deadlock patches into bundled<br>(Reported by Ross Beer)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26897" target="_blank">ASTERISK-26897</a>] - </li></ul></td><td></td><td>chan_sip: Security vulnerability with client code header<br>(Reported by Alex Villacís Lasso)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25974" target="_blank">ASTERISK-25974</a>] - </li></ul></td><td></td><td>Unused realtime MOH classes not purged on 'moh reload'<br>(Reported by Sébastien Couture)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26916" target="_blank">ASTERISK-26916</a>] - </li></ul></td><td></td><td>res_pjsip: Excessive refcount reached on transport ao2 object<br>(Reported by Ross Beer)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21721" target="_blank">ASTERISK-21721</a>] - </li></ul></td><td></td><td>SIP Failed to parse multiple Supported: headers<br>(Reported by Olle Johansson)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26915" target="_blank">ASTERISK-26915</a>] - </li></ul></td><td></td><td>chan_sip: Session Timers required but refused wrongly.<br>(Reported by Alexander Traud)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26363" target="_blank">ASTERISK-26363</a>] - </li></ul></td><td></td><td>res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code<br>(Reported by Yaacov Akiba Slama)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26896" target="_blank">ASTERISK-26896</a>] - </li></ul></td><td></td><td>Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT<br>(Reported by twisted)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26705" target="_blank">ASTERISK-26705</a>] - </li></ul></td><td></td><td>libasteriskssl.so not found when asterisk is installed for the 1st time<br>(Reported by George Joseph)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21009" target="_blank">ASTERISK-21009</a>] - </li></ul></td><td></td><td>xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client<br>(Reported by Marcello Ceschia)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25490" target="_blank">ASTERISK-25490</a>] - </li></ul></td><td></td><td>[patch]SDP crypto tag is validated incorrectly<br>(Reported by Joerg Sonnenberger)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26086" target="_blank">ASTERISK-26086</a>] - </li></ul></td><td></td><td>res_musiconhold: format option is not documented adequately<br>(Reported by Jens Bürger)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23996" target="_blank">ASTERISK-23996</a>] - </li></ul></td><td></td><td>No core dumps because of res_musiconhold chdir.<br>(Reported by Walter Doekes)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24712" target="_blank">ASTERISK-24712</a>] - </li></ul></td><td></td><td>xmpp: starttls problem causes connection spew<br>(Reported by Matthias Urlichs)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26814" target="_blank">ASTERISK-26814</a>] - </li></ul></td><td></td><td>pjproject_bundled build fails to download pjproject source when using cURL<br>(Reported by Gergely Dömsödi)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23510" target="_blank">ASTERISK-23510</a>] - </li></ul></td><td></td><td>JABBER_STATUS fails with improper code 7 for unavailable clients<br>(Reported by Anthony Critelli)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21855" target="_blank">ASTERISK-21855</a>] - </li></ul></td><td></td><td>Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available<br>(Reported by Jeremy Kister)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25622" target="_blank">ASTERISK-25622</a>] - </li></ul></td><td></td><td>WARNING for "JABBER: socket read error" should be more specific<br>(Reported by Sean Darcy)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26515" target="_blank">ASTERISK-26515</a>] - </li></ul></td><td></td><td>rtp_engine: Allocate RTP payloads on a per-session basis<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26818" target="_blank">ASTERISK-26818</a>] - </li></ul></td><td></td><td>cdr: Problem setting variables in h exten<br>(Reported by scgm11)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26875" target="_blank">ASTERISK-26875</a>] - </li></ul></td><td></td><td>app_mixmonitor: Recording out of sync when 183 but no RTP<br>(Reported by Aaron An)</td></tr>
</tbody></table>
<p>
<b>Improvements made in this release:</b><br>
-----------------------------------<br>
</p><table border="0">
<tbody><tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26088" target="_blank">ASTERISK-26088</a>] - </li></ul></td><td></td><td>Investigate heavy memory utilization by res_pjsip_pubsub<br>(Reported by Richard Mudgett)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26427" target="_blank">ASTERISK-26427</a>] - </li></ul></td><td></td><td>res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip<br>(Reported by Nir Simionovich (GreenfieldTech - Israel))</td></tr>
</tbody></table>
<p>
For a full list of changes in this release, please see the ChangeLog:<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0</a>
</p><p>
<b>Thank you for your continued support of Asterisk!</b><br>
</p><p></p><p></p><p></p><p></p><p></p><p></p></div>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></div></div>