The Asterisk Development Team has announced the release of Asterisk 13.10.0.

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

The release of Asterisk 13.10.0 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:

Improvements made in this release:

———————————–

* ASTERISK-26088 – Investigate heavy memory utilization by

     res_pjsip_pubsub (Reported by Richard Mudgett)

* ASTERISK-26011 – PJSIP: add “via_addr”, “via_port”,

     “call_id” to contacts (Reported by Alexei Gradinari)

* ASTERISK-25994 – res_pjsip: module load priority

     (Reported by Alexei Gradinari)

* ASTERISK-25931 – PJSIP: add “reg_server” to contacts. (Reported

     by Alexei Gradinari)

* ASTERISK-25835 – Authentication using ‘Username’ field from

     Digest (Reported by Ross Beer)

* ASTERISK-25930 – PJSIP: disable multi domain to improve realtime

     performace (Reported by Alexei Gradinari)

Bugs fixed in this release:

———————————–

* ASTERISK-26160 – pjsip: Updated->Reachable during qualify

     (Reported by Matt Jordan)

* ASTERISK-26177 – func_odbc: Database handle is kept when it

     should be released (Reported by Leandro Dardini)

* ASTERISK-26099 – res_pjsip_pubsub: Crash when sending request

     due to server timeout (Reported by Ross Beer)

* ASTERISK-26141 – res_fax: fax_v21_session_new leaks reference to

     v21_details (Reported by Corey Farrell)

* ASTERISK-26140 – res_rtp_asterisk: gcc 6 caught a

     self-comparison (Reported by George Joseph)

* ASTERISK-26138 – chan_unistim:  Under FreeBSD, chan_unistim

     generates a compile error (Reported by George Joseph)

* ASTERISK-26128 – Alembic scripts are failing (Reported by Mark

     Michelson)

* ASTERISK-26139 – test_res_pjsip_scheduler:  Compile failure if

     pjproject isn’t installed in a system location (Reported by

     George Joseph)

* ASTERISK-26130 – WebRTC: Should use latest DTLS version.

     (Reported by Alexander Traud)

* ASTERISK-26127 – res_pjsip_session: Crash due to race condition

     between res_pjsip_session unload and timer (Reported by Joshua

     Colp)

* ASTERISK-26083 – ARI: Announcer channels staying around after

     playback to a bridge is finished (Reported by Per Jensen)

* ASTERISK-26126 – leverage ‘bindaddr’ for TLS in

     http.conf (Reported by Alexander Traud)

* ASTERISK-26069 – Asterisk truncates To: header, dropping the

     closing ‘>’ (Reported by Vasil Kolev)

* ASTERISK-26097 – CLI: show maximum file descriptors

     (Reported by Alexander Traud)

* ASTERISK-25262 – Memory leak when a caller channel does multiple

     dials and CEL is enabled (Reported by Etienne Lessard)

* ASTERISK-26092 – [Segfault] in res_rtp_asterisk.c:4268 after

     Remotely bridged channels (Reported by Niklas Larsson)

* ASTERISK-26096 – res_hep: Crash when configuration file is

     missing (Reported by Niklas Larsson)

* ASTERISK-26089 – Invalid security events during boot using PJSIP

     Realtime (Reported by Scott Griepentrog)

* ASTERISK-26074 – res_odbc: Deadlock within UnixODBC (Reported by

     Ross Beer)

* ASTERISK-26054 – Asterisk crashes (core dump) (Reported by B.

     Davis)

* ASTERISK-24436 – Missing header in res/res_srtp.c when compiling

     against libsrtp-1.5.0 (Reported by Patrick Laimbock)

* ASTERISK-26091 – ar cru creates warning, instead use ar

     cr (Reported by Alexander Traud)

* ASTERISK-26070 – ari/channels:  Creating a local channel without

     an originator adds all audio formats to it’s capabilities

     (Reported by George Joseph)

* ASTERISK-26078 – core: Memory leak in logging (Reported by

     Etienne Lessard)

* ASTERISK-26065 – chan_pjsip: MWI NOTIFY contents not ordered

     properly (Reported by Ross Beer)

* ASTERISK-26063 – ${PJSIP_HEADER(read,Call-ID)} does not work –

     documentation needs clarification for when read/write is

     possible (Reported by Private Name)

* ASTERISK-25777 – data race in threadpool (Reported by Badalian

     Vyacheslav)

* ASTERISK-26038 – ‘make install’ doesn’t seem to install OS/X

     init files (Reported by Tzafrir Cohen)

* ASTERISK-26029 – parking: ast_parking_park_call should return

     parking_space instead of parking_exten (Reported by Diederik de

     Groot)

* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin

     Vandamme)

* ASTERISK-25941 – chan_pjsip: Crash on an immediate SIP final

     response (Reported by Javier Riveros )

* ASTERISK-26014 – res_sorcery_astdb: Make tolerant of unknown

     fields (Reported by Joshua Colp)

* ASTERISK-24986 – keepalive INFO packages ignored by asterisk

     (Reported by Ilya Trikoz)

* ASTERISK-26034 – T.38 passthrough problem behind firewall due to

     early nosignal packet (Reported by George Joseph)

* ASTERISK-26030 – call cut because of double Session-Expires

     header in re-invite after proxy authentication is required

     (Reported by George Joseph)

* ASTERISK-25964 – Outbound registrations created via ARI/push

     configuration do not clean up outbound registrations currently

     in flight (Reported by Matt Jordan)

* ASTERISK-26005 – res_pjsip: Multiple SIP messages are combined

     into 1 TCP packet (Reported by Ross Beer)

* ASTERISK-25352 – res_hep_rtcp correlation_id is different then

     res_hep (Reported by Kevin Scott Adams)

* ASTERISK-26008 – app_followme does not delete recorded name

     prompt (Reported by Tzafrir Cohen)

* ASTERISK-26007 – res_pjsip: Endpoints deleting early after

     upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

* ASTERISK-25990 – PJSIP TLS registration should respect

     client_uri scheme when generating Contact URI (Reported by

     Sebastian Damm)

* ASTERISK-25978 – res_pjsip_authenticator_digest: Should not use

     source port in nonce verification (Reported by Mark Michelson)

* ASTERISK-25993 – pjproject: Allow bundling to not require

     everything it does (Reported by Joshua Colp)

* ASTERISK-25956 – Compilation error in conditionally compiled

     code in config_options.c (Reported by Chris Trobridge)

* ASTERISK-25998 – file: Crash when using nativeformats (Reported

     by Joshua Colp)

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

     (Reported by Ross Beer)

* ASTERISK-25968 – pjproject_bundled:  Configure and make need to

     be re-tested (Reported by George Joseph)

* ASTERISK-24463 – Voicemail email address corrupt or not sent

     when message is in the process of being recorded during reload

     (Reported by John Campbell)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25961 – tests/channels/SIP/sip_tls_call: Sporadic crash

     when running test (Reported by Joshua Colp)

* ASTERISK-16115 – problem with ringinuse=no, queue

     members receive sometimes two calls (Reported by nik600)

* ASTERISK-25917 – app_voicemail: passwordlocation=spooldir

     only works if you manually add secret.conf yourself (Reported by

     Jonathan R. Rose)

* ASTERISK-25950 – SIP channel does not send PeerStatus

     events for autocreated peers (Reported by Kirill Katsnelson)

* ASTERISK-25954 – Manager QueueSummary and QueueStatus Actions

     are case sensitive to QueueName (Reported by Javier Acosta)

New Features made in this release:

———————————–

* ASTERISK-25904 – PJSIP: add contact.updated event (Reported by

     Alexei Gradinari)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0

Thank you for your continued support of Asterisk!

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