The Asterisk Development Team has announced the release of Asterisk 14.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.2.0 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-26558 - app_queue: add variable to know if the call is
not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
(Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to
configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
blacklisting host subnets that are not involved in RTP (Reported
by Michael Walton)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
by snuffy)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content
(Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus
negotiated but codec_opus not loaded. (Reported by Richard
Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory
leak. (Reported by Richard Mudgett)
* ASTERISK-26556 - manager: AMI version report same in Ast 13 &
14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality
when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit
IPv6 transport configured (Reported by Joshua Colp)
* ASTERISK-26468 - ari: Bridge events stop working after this
sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George
Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when
building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state (Reported by Joshua
Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
payload types. (Reported by Alexander Traud)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when
publishing, in publisher_client_send at
res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
reg. retry 403" in "sip show settings" (Reported by Sergey
Grachev)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
to maximum (Reported by Joshua Colp)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented
(Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space
when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even
with no active calls. (Reported by Harley Peters)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport
in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components
option of tar which isn't supported in older versions (Reported
by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains
hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
(Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
installations. (Reported by Alexander Traud)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already
disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26421 - Segmentation Fault with ARI originate into
mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return
prompt. (Reported by John Kiniston)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
Tzafrir Cohen)
* ASTERISK-26462 - [patch] app_queue: While using queues with
realtime, setting back to an empty context doesn't stop the exit
key usage (Reported by Leandro Dardini)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
Kayode)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
New Features made in this release:
-----------------------------------
* ASTERISK-26595 - ARI: Add the ability to control the source of
video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26492 - ARI: Add ability to specify channel variables
on websocket events (Reported by Mark Michelson)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
events (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0
Thank you for your continued support of Asterisk!
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