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The Asterisk Development Team would like to announce the release of Asterisk 13.17.0.<br>
This release is available for immediate download at<br>
<a href='http://downloads.asterisk.org/pub/telephony/asterisk'>http://downloads.asterisk.org/pub/telephony/asterisk</a>
<p>
The release of Asterisk 13.17.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
<p>
<b>Thank you!</b><br>
<p>
The following issues are resolved in this release:<br>
<p>
<b>Bugs fixed in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27108'>ASTERISK-27108</a>] - <td><td>Crash using 'data get' CLI command<br>(Reported by Sean Bright)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27106'>ASTERISK-27106</a>] - <td><td>[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27100'>ASTERISK-27100</a>] - <td><td>channel: ast_waitfordigit_full fails to clear flag in an error branch.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27090'>ASTERISK-27090</a>] - <td><td>PJSIP: Deadlock using TCP transport<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25665'>ASTERISK-25665</a>] - <td><td>Duplicate logging in queue log for EXITEMPTY events<br>(Reported by Ove Aursand)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27065'>ASTERISK-27065</a>] - <td><td>call hangup after leaving app_queue<br>(Reported by Marek Cervenka)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26978'>ASTERISK-26978</a>] - <td><td>rtp: Crash in ast_rtp_codecs_payload_code()<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27074'>ASTERISK-27074</a>] - <td><td>core_local: local channel data not being properly unref'ed and unlocked<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27075'>ASTERISK-27075</a>] - <td><td>bridge: stuck channel(s) after failed attended transfer<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24052'>ASTERISK-24052</a>] - <td><td>app_voicemail reloads result in leaked IMAP sockets.<br>(Reported by Louis Jocelyn Paquet)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27060'>ASTERISK-27060</a>] - <td><td>Comment typo format_g729.c<br>(Reported by Matthew Fredrickson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27026'>ASTERISK-27026</a>] - <td><td>res_ari: Crash when no ari.conf configuration file exists<br>(Reported by Ronald Raikes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27041'>ASTERISK-27041</a>] - <td><td>Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br>(Reported by Frederic LE FOLL)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27057'>ASTERISK-27057</a>] - <td><td>Seg Fault in ast_sorcery_object_get_id at sorcery.c<br>(Reported by Ryan Smith)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27024'>ASTERISK-27024</a>] - <td><td>nat/external_media settings ignored in 14.4.1<br>(Reported by Christopher van de Sande)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27046'>ASTERISK-27046</a>] - <td><td>res_pjsip_transport_websocket: segfault in get_write_timeout<br>(Reported by Jørgen H)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27022'>ASTERISK-27022</a>] - <td><td>res_rtp_asterisk: Incorrect SSRC change for RTCP component<br>(Reported by Michael Walton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26923'>ASTERISK-26923</a>] - <td><td>bridging: T.38 request is lost when channels are added to bridge<br>(Reported by Torrey Searle)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27053'>ASTERISK-27053</a>] - <td><td>res_pjsip_refer/session: Calls dropped during transfer<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27052'>ASTERISK-27052</a>] - <td><td>Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br>(Reported by alex)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27039'>ASTERISK-27039</a>] - <td><td>chan_pjsip: Device state is idle when channel from endpoint is in early media<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26996'>ASTERISK-26996</a>] - <td><td>chan_pjsip: Flipping between codecs<br>(Reported by Michael Maier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26281'>ASTERISK-26281</a>] - <td><td>chan_pjsip would send INVITE to 'Unreachable' endpoints<br>(Reported by Jacek Konieczny)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26973'>ASTERISK-26973</a>] - <td><td>bridge: Crash when freeing frame and snooping<br>(Reported by Michel R. Vaillancourt)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-19291'>ASTERISK-19291</a>] - <td><td>Background in realtime<br>(Reported by Andrew Nowrot)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27025'>ASTERISK-27025</a>] - <td><td>channel / meetme: Fix missing parentheses<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27021'>ASTERISK-27021</a>] - <td><td>GET /recordings/stored returns 500 Internal Server Error<br>(Reported by Tim Morgan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24858'>ASTERISK-24858</a>] - <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)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23951'>ASTERISK-23951</a>] - <td><td> Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25294'>ASTERISK-25294</a>] - <td><td>srtp's crypto_get_random deprecated<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23839'>ASTERISK-23839</a>] - <td><td>AGI - RECORD FILE - documentation doesn't describe BEEP argument<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22432'>ASTERISK-22432</a>] - <td><td>Async AGI crashes Asterisk when issuing "set variable" command without args<br>(Reported by Antoine Pitrou)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25662'>ASTERISK-25662</a>] - <td><td>Malformed AGI 520 Usage response<br>(Reported by Tony Mountifield)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25101'>ASTERISK-25101</a>] - <td><td>DTLS configuration can not be specified in the general section - documentation<br>(Reported by Ben Langfeld)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27008'>ASTERISK-27008</a>] - <td><td>res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br>(Reported by John Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26399'>ASTERISK-26399</a>] - <td><td>app_queue: Agent not called when caller is parked<br>(Reported by wushumasters)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26400'>ASTERISK-26400</a>] - <td><td>app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26715'>ASTERISK-26715</a>] - <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)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26975'>ASTERISK-26975</a>] - <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)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27012'>ASTERISK-27012</a>] - <td><td>app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br>(Reported by Robert Mordec)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26979'>ASTERISK-26979</a>] - <td><td>res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br>(Reported by Javier Riveros )</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26982'>ASTERISK-26982</a>] - <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)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26964'>ASTERISK-26964</a>] - <td><td>res_pjsip_session: Wrong From on reinvite when request and To URI differ<br>(Reported by Yasin CANER)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26789'>ASTERISK-26789</a>] - <td><td>Audit manipulation of channel flags without locks<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26333'>ASTERISK-26333</a>] - <td><td>Problems with Blind Transfer, PJSIP (Aastra 6869i)<br>(Reported by Matthias Binder)</li></td></tr>
</table>
<p>
<b>Improvements made in this release:</b><br>
-----------------------------------<br>
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<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26230'>ASTERISK-26230</a>] - <td><td>[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27043'>ASTERISK-27043</a>] - <td><td>Core/BuildSystem: Add defines to fix build with LibreSSL<br>(Reported by Guido Falsi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27042'>ASTERISK-27042</a>] - <td><td>Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br>(Reported by Guido Falsi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26419'>ASTERISK-26419</a>] - <td><td>audiohooks: Remove redundant codec translations when using audiohooks<br>(Reported by Michael Walton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26976'>ASTERISK-26976</a>] - <td><td>libsrtp-2.x.x support<br>(Reported by Alex)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26124'>ASTERISK-26124</a>] - <td><td>res_agi: Set audio format for EAGI audio stream<br>(Reported by John Fawcett)</li></td></tr>
</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-13.17.0'>http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0</a>
<p>
<b>Thank you for your continued support of Asterisk!</b><br>
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