<html><head></head><body>
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.31.0.<br>
This release candidate is available for immediate download at <br>
<a href='https://downloads.asterisk.org/pub/telephony/asterisk'>https://downloads.asterisk.org/pub/telephony/asterisk</a>
<p>
The release of Asterisk 13.31.0-rc1 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 candidate:<br>
<p>
<b>New Features made 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-17491'>ASTERISK-17491</a>] - <td><td>CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything<br>(Reported by candrews)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28639'>ASTERISK-28639</a>] - <td><td>res_pjsip_endpoint_identifier_ip: Add ability to match on source port<br>(Reported by Sean Bright)</li></td></tr>
</table>
<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-28677'>ASTERISK-28677</a>] - <td><td>CDR billsec is always 0 for transferred calls<br>(Reported by Maciej Michno)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28706'>ASTERISK-28706</a>] - <td><td>silk 24hHz doesn't show up in 'core show translation' output<br>(Reported by Sean Bright)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24484'>ASTERISK-24484</a>] - <td><td>Update documentation for statsd module - usage requirements unclear<br>(Reported by Dan Jenkins)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28702'>ASTERISK-28702</a>] - <td><td>chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40<br>(Reported by Andrew Siplas)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28695'>ASTERISK-28695</a>] - <td><td>core: minmemfree watermark uses free RAM, not available RAM<br>(Reported by Kevin Flyn)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28693'>ASTERISK-28693</a>] - <td><td>chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan<br>(Reported by Frank Matano)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23739'>ASTERISK-23739</a>] - <td><td>[patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used<br>(Reported by Stas Kobzar)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27622'>ASTERISK-27622</a>] - <td><td>empty voicemail.conf required for ARA (realtime) voicemail to leave message<br>(Reported by Jim Van Meggelen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28349'>ASTERISK-28349</a>] - <td><td>Pause reason not reported in QueueMember AMI event<br>(Reported by Niksa Baldun)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21794'>ASTERISK-21794</a>] - <td><td>CLI command 'realtime update2' syntax failure when using according to usage help<br>(Reported by Cedric BASSAGET)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25429'>ASTERISK-25429</a>] - <td><td>res_pjsip_endpoint_identifier_ip: Document support for hostnames<br>(Reported by Joshua C. Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27775'>ASTERISK-27775</a>] - <td><td>res_pjsip_notify: Multiple Event headers can be present instead of just one<br>(Reported by AvayaXAsterisk)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28682'>ASTERISK-28682</a>] - <td><td>app_record: Lack of `beep` audio file causes application to return error and hangup<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28507'>ASTERISK-28507</a>] - <td><td>Wiki docs missing for MessageWaiting<br>(Reported by David M. Lee)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27759'>ASTERISK-27759</a>] - <td><td>res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number<br>(Reported by Bryan Nelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28605'>ASTERISK-28605</a>] - <td><td>chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X<br>(Reported by Dirk Wendland)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28633'>ASTERISK-28633</a>] - <td><td>stasis bridge topic leak<br>(Reported by Joeran Vinzens)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28492'>ASTERISK-28492</a>] - <td><td>pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group<br>(Reported by Jean-Denis Girard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27243'>ASTERISK-27243</a>] - <td><td>contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax<br>(Reported by Richard Kenner)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28497'>ASTERISK-28497</a>] - <td><td>func_odbc: truncating Unicode string on readsql<br>(Reported by Boris P. Korzun)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28647'>ASTERISK-28647</a>] - <td><td>chan_sip: RTP frames not transmitted after emitting a COLP<br>(Reported by Jean Aunis - Prescom)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28667'>ASTERISK-28667</a>] - <td><td>Asterisk ignores parsing of config files if a Byte order mark is present<br>(Reported by Robin Leffmann)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28664'>ASTERISK-28664</a>] - <td><td>"trustrpid" is misspelled in sip_to_pjsip.py<br>(Reported by Pascal Cadotte Michaud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28663'>ASTERISK-28663</a>] - <td><td>jansson: Support old versions<br>(Reported by Joshua C. Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28636'>ASTERISK-28636</a>] - <td><td>app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.<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-28604'>ASTERISK-28604</a>] - <td><td>app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28660'>ASTERISK-28660</a>] - <td><td>res_fax: wrap Asterisk initiated negotiation with config option<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28628'>ASTERISK-28628</a>] - <td><td>Debian 10.2: Warning when app_voicemail is compiling<br>(Reported by Stanislav Abramenkov)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28626'>ASTERISK-28626</a>] - <td><td>Missing arguments in PJSIP_CONTACT function documentation<br>(Reported by Pascal Cadotte Michaud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28651'>ASTERISK-28651</a>] - <td><td>chan_sip logs errors on tx to non-existent TCP connections<br>(Reported by Jaco Kroon)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28502'>ASTERISK-28502</a>] - <td><td>chan_pjsip incorrectly re-writes REGISTER 200 Response Contact<br>(Reported by Ross Beer)</li></td></tr>
</table>
<p>
<b>Improvements made 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-28710'>ASTERISK-28710</a>] - <td><td>Should be able to disable the /httpstatus URI in the built-in HTTP server<br>(Reported by Sean Bright)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28638'>ASTERISK-28638</a>] - <td><td>Simplify dialplan for Dial, Page, and ChanIsAvail<br>(Reported by cmaj)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28673'>ASTERISK-28673</a>] - <td><td>GET FULL VARIABLE documentation clarification<br>(Reported by Jonathan Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28658'>ASTERISK-28658</a>] - <td><td>app_confbridge: Add support for setting maximum sample rate<br>(Reported by Joshua C. Colp)</li></td></tr>
</table>
<p>
For a full list of changes in this release candidate, please see the ChangeLog:<br>
<a href='https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.31.0-rc1'>https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.31.0-rc1</a>
<p>
<b>Thank you for your continued support of Asterisk!</b><br>
</body></html>