Asterisk Development Team
2016-Jul-21 17:38 UTC
[asterisk-users] Asterisk 13.10.0 Now Available
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 - [patch]PJSIP: add "via_addr",
"via_port",
"call_id" to contacts (Reported by Alexei Gradinari)
* ASTERISK-25994 - [patch]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 - [patch] 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 - [patch] 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 - [patch] 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 - [patch] 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 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25950 - [patch]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!
21.7.2016, 20:38, Asterisk Development Team kirjoitti:> Bugs fixed in this release: > ----------------------------------- > * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. > (Reported by Alexander Traud)Now it's possible to use dtls_cipher settings such like: dtls_cipher=ALL:!SSLv3 or dtls_cipher=HIGH:!SSLv3 Thank you! Best, Teijo
On Thu, Jul 21, 2016 at 4:18 PM, Teijo <g.aloitus at gmail.com> wrote:> > > 21.7.2016, 20:38, Asterisk Development Team kirjoitti: >> >> Bugs fixed in this release: >> ----------------------------------- >> * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. >> (Reported by Alexander Traud) > > > Now it's possible to use dtls_cipher settings such like: > > dtls_cipher=ALL:!SSLv3 > or > dtls_cipher=HIGH:!SSLv3 > > Thank you! >I'll echo that sentiment - Alexander has done a lot of work recently to improve Asterisk's support of available ciphers both in DTLS and SRTP. Thanks Alexander! -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org