The Asterisk Development Team has announced the release of Asterisk 13.9.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.9.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:

———————————–

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin

     Vandamme)

* ASTERISK-25927 – Removed option “registertrying” is still

     documented in sip.conf.sample (Reported by Etienne Lessard)

* ASTERISK-25947 – Protocol transfers to stasis applications are

     missing the StasisStart with the replace_channel object.

     (Reported by Richard Mudgett)

* ASTERISK-24649 – Pushing of channel into bridge fails; Stasis

     fails to get app name (Reported by John Bigelow)

* ASTERISK-24782 – StasisEnd event not present for channel that

     was swapped out for another after completing attended transfer

     (Reported by John Bigelow)

* ASTERISK-25942 – res_pjsip_caller_id: Transfer results in mixed

     ConnectedLine information (Reported by George Joseph)

* ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

     thread (Reported by Joshua Colp)

* ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

     not raised (Reported by Joshua Colp)

* ASTERISK-25934 – chan_sip should not require sipregs or

     updateable sippeers table unless rt (Reported by Jaco Kroon)

* ASTERISK-25888 – Frequent segfaults in function can_ring_entry()

     of app_queue.c (Reported by Sébastien Couture)

* ASTERISK-25796 – res_pjsip: DOS/Crash when TCP/TLS sockets

     exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George

     Joseph)

* ASTERISK-25707 – Long contact URIs or hostnames can crash

     pjproject/Asterisk under certain conditions (Reported by George

     Joseph)

* ASTERISK-25123 – Bracketed IPv6 Contact header parameter

     unparsable with Asterisk/PJSIP (Reported by Anthony Messina)

* ASTERISK-25874 – app_voicemail: Stack buffer overflow in

     test_voicemail_notify_endl (Reported by Badalian Vyacheslav)

* ASTERISK-25912 – chan_local passes AST_CONTROL_PVT_CAUSE_CODE

     without adding them to the local hangupcauses via

     ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)

* ASTERISK-25885 – res_pjsip: Race condition between adding

     contact and automatic expiration (Reported by Joshua Colp)

* ASTERISK-25910 – pjproject:  Via headers are not parsed when

     “received” contains an IPv6 address (Reported by George Joseph)

* ASTERISK-25890 – Asterisk 13.8.0 alembic database update fails

     (Reported by Harley Peters)

* ASTERISK-25894 – webrtc video broken due to missing

     marker bits in RTP streams (Reported by Jacek Konieczny)

* ASTERISK-25854 – No audio after HOLD/RESUME – incorrect

     a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)

* ASTERISK-25873 – res_pjsip: Bundled pjproject: compile error,

     cannot find -lasteriskpj (Reported by Hans van Eijsden)

* ASTERISK-25882 – ARI: Crash can occur due to race condition when

     attempting to operate on a hung up channel (Part 2) (Reported by

     Richard Mudgett)

* ASTERISK-25867 – Video delay on app_echo (Reported by

     Jacek Konieczny)

* ASTERISK-24605 – res_parking option parkeddynamic does not work

     with the core Features ‘parkcall’ (DTMF initiated parking)

     (Reported by Philip Correia)

* ASTERISK-25826 – PJSIP / Sorcery slow load from realtime

     (Reported by Ross Beer)

* ASTERISK-24596 – Unclear how to use Park application with

     res_parking ‘parkeddynamic’ enabled. Documentation? (Reported by

     Philip Correia)

* ASTERISK-24543 – Asterisk 13 responds to SIP Invite with all

     possible codecs configured for peer as opposed to intersection

     of configured codecs and offered codecs (Reported by Taylor

     Hawkes)

* ASTERISK-25825 – Crashes during shutdown when running CLI

     commands (Reported by Mark Michelson)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25510 – Log to syslog failing (Reported by

     Michael Newton)

* ASTERISK-25857 – func_aes: incorrect use of strlen() leads to

     data corruption (Reported by Gianluca Merlo)

Improvements made in this release:

———————————–

* ASTERISK-25865 – Message-Account Missing From PJSIP MWI

     (Reported by Ross Beer)

* ASTERISK-25444 – Music On Hold Warning misleading

     (Reported by Conrad de Wet)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0

Thank you for your continued support of Asterisk!

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