[asterisk-dev] Asterisk 13.14.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Mon Feb 6 12:56:32 CST 2017
The Asterisk Development Team has announced the first release candidate of
Asterisk 13.14.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.14.0-rc1 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 candidate:
New Features made in this release:
-----------------------------------
* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26704 - res_odbc.conf contains deprecated
configuration: 'pooling', 'shared_connections', 'limit', and
'idlecheck' options were replaced by 'max_connections'.
(Reported by Anthony Messina)
* ASTERISK-21094 - MixMonitorMute mutes through stream if already
slinear (e.g. Originate) (Reported by David Woolley)
* ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
hung up via ARI (Reported by Tom Pawelek)
* ASTERISK-26632 - core: Possibility of a frame "imbalance"
leading to stuck channels. (Reported by Mark Michelson)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
(Reported by Dmitriy)
* ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
request to endpoint (Reported by Ross Beer)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-26754 - build_tools: make_build_h does not handle \ in
user name (Reported by Kirill Katsnelson)
* ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
fwrite() returned error: Broken pipe" (Reported by Kirill
Katsnelson)
* ASTERISK-26755 - app_queue: Random queues disappear on "core
reload queue all" (Reported by Kirill Katsnelson)
* ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
after match in .conf has no effect (Reported by Michael Maier)
* ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
for SRV (Reported by Joshua Colp)
* ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
does not work. (Reported by Richard Mudgett)
* ASTERISK-26740 - voicemail API test: uses varlibdir instead of
datadir for a sound file (Reported by Tzafrir Cohen)
* ASTERISK-26739 - voicemail API test: confuses expected and
actual values (Reported by Tzafrir Cohen)
* ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
sorcery memory cache populate (Reported by Ustinov Artem)
* ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
(rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
(Reported by Aaron An)
* ASTERISK-26672 - Crash when setting remote address on RTP
instance (Reported by Richard Mudgett)
* ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
(Reported by Alexander Traud)
* ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
(Reported by Alexander Traud)
* ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
function around masquerade (Reported by Joshua Colp)
* ASTERISK-26684 - res_pjsip: Various issues with compact SIP
headers (Reported by Joshua Elson)
* ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
Headers Enabled (Reported by JoshE)
* ASTERISK-26621 - app_queue: Queue application does not ring
members with Local interface (Reported by Jonas Kellens)
* ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
MWI wasn't registered (Reported by Michael Kuron)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-26653 - pjproject_bundled doesn't verify already
downloaded tarballs (Reported by George Joseph)
* ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
setting up new calls (Reported by Walter Doekes)
* ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
line (Reported by Jørgen H)
* ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
aors (Reported by George Joseph)
* ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not
Exist when transaction branch parameter contains "_" (Reported
by Juris Breicis)
* ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without
IPv6 (Reported by Guido Falsi)
* ASTERISK-24330 - Requirement for 'wss' value in Contact header
transport parameter on inbound traffic violates RFC7118
(Reported by Marek Cervenka)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen)
* ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP
(Reported by Hector Royo Concepcion)
* ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to
tlscertfile, tlsciphers, etc. (Reported by Michael Kuron)
* ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec
to receiving codec when asymmetric_rtp_codec=no (Reported by
Alexei Gradinari)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
Improvements made in this release:
-----------------------------------
* ASTERISK-23828 - pjsip - Need a command to list active SIP
subscriptions (Reported by Rusty Newton)
* ASTERISK-26527 - Testsuite: increase timeout to check "core
fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav)
* ASTERISK-26624 - res_calendar_caldav: Add support for gmail
(Reported by Eduardo Scudeller Libardi)
* ASTERISK-26562 - app_controlplayback: Transmit Silence on
ControlPlayback pause (Reported by Mikheili Dautashvili)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0-rc1
Thank you for your continued support of Asterisk!
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