<div dir="ltr"><div class="gmail_quote"><br>The Asterisk Development Team has announced the release of Asterisk 13.9.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 13.9.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following are the issues resolved in this release:<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-25963 - func_odbc requires reconnect checks for stale<br>
connections (Reported by Ross Beer)<br>
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by<br>
Dmitriy Serov)<br>
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement<br>
LAST_INSERT_ID() always returns zero. (Reported by Edwin<br>
Vandamme)<br>
* ASTERISK-25927 - Removed option "registertrying" is still<br>
documented in sip.conf.sample (Reported by Etienne Lessard)<br>
* ASTERISK-25947 - Protocol transfers to stasis applications are<br>
missing the StasisStart with the replace_channel object.<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis<br>
fails to get app name (Reported by John Bigelow)<br>
* ASTERISK-24782 - StasisEnd event not present for channel that<br>
was swapped out for another after completing attended transfer<br>
(Reported by John Bigelow)<br>
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed<br>
ConnectedLine information (Reported by George Joseph)<br>
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP<br>
thread (Reported by Joshua Colp)<br>
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events<br>
not raised (Reported by Joshua Colp)<br>
* ASTERISK-25934 - chan_sip should not require sipregs or<br>
updateable sippeers table unless rt (Reported by Jaco Kroon)<br>
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()<br>
of app_queue.c (Reported by Sébastien Couture)<br>
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets<br>
exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George<br>
Joseph)<br>
* ASTERISK-25707 - Long contact URIs or hostnames can crash<br>
pjproject/Asterisk under certain conditions (Reported by George<br>
Joseph)<br>
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter<br>
unparsable with Asterisk/PJSIP (Reported by Anthony Messina)<br>
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in<br>
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)<br>
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE<br>
without adding them to the local hangupcauses via<br>
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)<br>
* ASTERISK-25885 - res_pjsip: Race condition between adding<br>
contact and automatic expiration (Reported by Joshua Colp)<br>
* ASTERISK-25910 - pjproject: Via headers are not parsed when<br>
"received" contains an IPv6 address (Reported by George Joseph)<br>
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails<br>
(Reported by Harley Peters)<br>
* ASTERISK-25894 - [patch] webrtc video broken due to missing<br>
marker bits in RTP streams (Reported by Jacek Konieczny)<br>
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect<br>
a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)<br>
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,<br>
cannot find -lasteriskpj (Reported by Hans van Eijsden)<br>
* ASTERISK-25882 - ARI: Crash can occur due to race condition when<br>
attempting to operate on a hung up channel (Part 2) (Reported by<br>
Richard Mudgett)<br>
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by<br>
Jacek Konieczny)<br>
* ASTERISK-24605 - res_parking option parkeddynamic does not work<br>
with the core Features 'parkcall' (DTMF initiated parking)<br>
(Reported by Philip Correia)<br>
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime<br>
(Reported by Ross Beer)<br>
* ASTERISK-24596 - Unclear how to use Park application with<br>
res_parking 'parkeddynamic' enabled. Documentation? (Reported by<br>
Philip Correia)<br>
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all<br>
possible codecs configured for peer as opposed to intersection<br>
of configured codecs and offered codecs (Reported by Taylor<br>
Hawkes)<br>
* ASTERISK-25825 - Crashes during shutdown when running CLI<br>
commands (Reported by Mark Michelson)<br>
* ASTERISK-25407 - Asterisk fails to log to multiple syslog<br>
destinations (Reported by Elazar Broad)<br>
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by<br>
Michael Newton)<br>
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to<br>
data corruption (Reported by Gianluca Merlo)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-25865 - Message-Account Missing From PJSIP MWI<br>
(Reported by Ross Beer)<br>
* ASTERISK-25444 - [patch]Music On Hold Warning misleading<br>
(Reported by Conrad de Wet)<br>
<br>
For a full list of changes in this release, please see the ChangeLog:<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
_____________________________________________________________________<br>
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