<div dir="ltr"><div class="gmail_quote"><br><div>
The Asterisk Development Team would like to announce the release of Asterisk 14.6.0.<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 14.6.0 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-27108" target="_blank">ASTERISK-27108</a>] - </li></ul></td><td></td><td>Crash using 'data get' CLI command<br>(Reported by Sean Bright)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27106" target="_blank">ASTERISK-27106</a>] - </li></ul></td><td></td><td>[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br>(Reported by Alexander Traud)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27100" target="_blank">ASTERISK-27100</a>] - </li></ul></td><td></td><td>channel: ast_waitfordigit_full fails to clear flag in an error branch.<br>(Reported by Corey Farrell)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27090" target="_blank">ASTERISK-27090</a>] - </li></ul></td><td></td><td>PJSIP: Deadlock using TCP transport<br>(Reported by Richard Mudgett)</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-27065" target="_blank">ASTERISK-27065</a>] - </li></ul></td><td></td><td>call hangup after leaving app_queue<br>(Reported by Marek Cervenka)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26978" target="_blank">ASTERISK-26978</a>] - </li></ul></td><td></td><td>rtp: Crash in ast_rtp_codecs_payload_code()<br>(Reported by Ross Beer)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24052" target="_blank">ASTERISK-24052</a>] - </li></ul></td><td></td><td>app_voicemail reloads result in leaked IMAP sockets.<br>(Reported by Louis Jocelyn Paquet)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27074" target="_blank">ASTERISK-27074</a>] - </li></ul></td><td></td><td>core_local: local channel data not being properly unref'ed and unlocked<br>(Reported by Kevin Harwell)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27075" target="_blank">ASTERISK-27075</a>] - </li></ul></td><td></td><td>bridge: stuck channel(s) after failed attended transfer<br>(Reported by Kevin Harwell)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27060" target="_blank">ASTERISK-27060</a>] - </li></ul></td><td></td><td>Comment typo format_g729.c<br>(Reported by Matthew Fredrickson)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27041" target="_blank">ASTERISK-27041</a>] - </li></ul></td><td></td><td>Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br>(Reported by Frederic LE FOLL)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27026" target="_blank">ASTERISK-27026</a>] - </li></ul></td><td></td><td>res_ari: Crash when no ari.conf configuration file exists<br>(Reported by Ronald Raikes)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27057" target="_blank">ASTERISK-27057</a>] - </li></ul></td><td></td><td>Seg Fault in ast_sorcery_object_get_id at sorcery.c<br>(Reported by Ryan Smith)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27024" target="_blank">ASTERISK-27024</a>] - </li></ul></td><td></td><td>nat/external_media settings ignored in 14.4.1<br>(Reported by Christopher van de Sande)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27046" target="_blank">ASTERISK-27046</a>] - </li></ul></td><td></td><td>res_pjsip_transport_websocket: segfault in get_write_timeout<br>(Reported by Jørgen H)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27022" target="_blank">ASTERISK-27022</a>] - </li></ul></td><td></td><td>res_rtp_asterisk: Incorrect SSRC change for RTCP component<br>(Reported by Michael Walton)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923" target="_blank">ASTERISK-26923</a>] - </li></ul></td><td></td><td>bridging: T.38 request is lost when channels are added to bridge<br>(Reported by Torrey Searle)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053" target="_blank">ASTERISK-27053</a>] - </li></ul></td><td></td><td>res_pjsip_refer/session: Calls dropped during transfer<br>(Reported by Kevin Harwell)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27052" target="_blank">ASTERISK-27052</a>] - </li></ul></td><td></td><td>Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br>(Reported by alex)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27039" target="_blank">ASTERISK-27039</a>] - </li></ul></td><td></td><td>chan_pjsip: Device state is idle when channel from endpoint is in early media<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26996" target="_blank">ASTERISK-26996</a>] - </li></ul></td><td></td><td>chan_pjsip: Flipping between codecs<br>(Reported by Michael Maier)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26281" target="_blank">ASTERISK-26281</a>] - </li></ul></td><td></td><td>chan_pjsip would send INVITE to 'Unreachable' endpoints<br>(Reported by Jacek Konieczny)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973" target="_blank">ASTERISK-26973</a>] - </li></ul></td><td></td><td>bridge: Crash when freeing frame and snooping<br>(Reported by Michel R. Vaillancourt)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19291" target="_blank">ASTERISK-19291</a>] - </li></ul></td><td></td><td>Background in realtime<br>(Reported by Andrew Nowrot)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025" target="_blank">ASTERISK-27025</a>] - </li></ul></td><td></td><td>channel / meetme: Fix missing parentheses<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27021" target="_blank">ASTERISK-27021</a>] - </li></ul></td><td></td><td>GET /recordings/stored returns 500 Internal Server Error<br>(Reported by Tim Morgan)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858" target="_blank">ASTERISK-24858</a>] - </li></ul></td><td></td><td>[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br>(Reported by Frankie Chin)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23951" target="_blank">ASTERISK-23951</a>] - </li></ul></td><td></td><td> Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br>(Reported by Tzafrir Cohen)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25294" target="_blank">ASTERISK-25294</a>] - </li></ul></td><td></td><td>srtp's crypto_get_random deprecated<br>(Reported by Tzafrir Cohen)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839" target="_blank">ASTERISK-23839</a>] - </li></ul></td><td></td><td>AGI - RECORD FILE - documentation doesn't describe BEEP argument<br>(Reported by Rusty Newton)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22432" target="_blank">ASTERISK-22432</a>] - </li></ul></td><td></td><td>Async AGI crashes Asterisk when issuing "set variable" command without args<br>(Reported by Antoine Pitrou)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25662" target="_blank">ASTERISK-25662</a>] - </li></ul></td><td></td><td>Malformed AGI 520 Usage response<br>(Reported by Tony Mountifield)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27008" target="_blank">ASTERISK-27008</a>] - </li></ul></td><td></td><td>res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br>(Reported by John Harris)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399" target="_blank">ASTERISK-26399</a>] - </li></ul></td><td></td><td>app_queue: Agent not called when caller is parked<br>(Reported by wushumasters)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26400" target="_blank">ASTERISK-26400</a>] - </li></ul></td><td></td><td>app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br>(Reported by Etienne Lessard)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26715" target="_blank">ASTERISK-26715</a>] - </li></ul></td><td></td><td>app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel<br>(Reported by David Brillert)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26975" target="_blank">ASTERISK-26975</a>] - </li></ul></td><td></td><td>app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call<br>(Reported by Lorne Gaetz)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27012" target="_blank">ASTERISK-27012</a>] - </li></ul></td><td></td><td>app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br>(Reported by Robert Mordec)</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-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-26964" target="_blank">ASTERISK-26964</a>] - </li></ul></td><td></td><td>res_pjsip_session: Wrong From on reinvite when request and To URI differ<br>(Reported by Yasin CANER)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26789" target="_blank">ASTERISK-26789</a>] - </li></ul></td><td></td><td>Audit manipulation of channel flags without locks<br>(Reported by Joshua Colp)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26333" target="_blank">ASTERISK-26333</a>] - </li></ul></td><td></td><td>Problems with Blind Transfer, PJSIP (Aastra 6869i)<br>(Reported by Matthias Binder)</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-26230" target="_blank">ASTERISK-26230</a>] - </li></ul></td><td></td><td>[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br>(Reported by Alexei Gradinari)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27043" target="_blank">ASTERISK-27043</a>] - </li></ul></td><td></td><td>Core/BuildSystem: Add defines to fix build with LibreSSL<br>(Reported by Guido Falsi)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-27042" target="_blank">ASTERISK-27042</a>] - </li></ul></td><td></td><td>Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br>(Reported by Guido Falsi)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419" target="_blank">ASTERISK-26419</a>] - </li></ul></td><td></td><td>audiohooks: Remove redundant codec translations when using audiohooks<br>(Reported by Michael Walton)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26976" target="_blank">ASTERISK-26976</a>] - </li></ul></td><td></td><td>libsrtp-2.x.x support<br>(Reported by Alex)</td></tr>
<tr><td valign="top" nowrap><ul><li>[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26124" target="_blank">ASTERISK-26124</a>] - </li></ul></td><td></td><td>res_agi: Set audio format for EAGI audio stream<br>(Reported by John Fawcett)</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-14.6.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0</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>
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