The Asterisk Development Team would like to announce the first
release candidate of Asterisk 14.6.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.6.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27074 - core_local: local channel data not being
properly unref'ed and unlocked
(Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed
attended transfer
(Reported by Kevin Harwell)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported
by Matthew Fredrickson)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by
Frederic LE FOLL)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
RTCP component
(Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels
are added to bridge
(Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
during transfer
(Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow
network
(Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when
channel from endpoint is in early media
(Reported by
Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to
'Unreachable' endpoints
(Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and
snooping
(Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime
(Reported by
Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal
Server Error
(Reported by Tim Morgan)
* 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-23951 - Asterisk attempts and fails to build
format_mp3 even if mp3lib was not downloaded
(Reported by
Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
describe BEEP argument
(Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
variable" command without args
(Reported by Antoine
Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
fmtp optional parameters have a space
(Reported by John
Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is
parked
(Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called
after AMI "Redirect" action for queues with wrapuptime
(Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new
calls after doing a transfer if wrapuptime = greater than 0 and
using Local channel
(Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
agents not to receive queue calls after transfer queue call
(Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
not play user name recording while leaving
(Reported by
Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier
Riveros )
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
Information Requests made in this release:
-----------------------------------
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by
Alex)
Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei
Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
with LibreSSL
(Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on
FreeBSD due to missing crypt.h file
(Reported by Guido
Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec
translations when using audiohooks
(Reported by Michael
Walton)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0-rc1
Thank you for your continued support of Asterisk!
--
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