The Asterisk Development Team would like to announce the release of
Asterisk 14.6.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.6.0 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:

*Bugs fixed in this release:*
-----------------------------------

   - [ASTERISK-27108
   <https://issues.asterisk.org/jira/browse/ASTERISK-27108>] -

Crash using 'data get' CLI command
(Reported by Sean Bright)

   - [ASTERISK-27106
   <https://issues.asterisk.org/jira/browse/ASTERISK-27106>] -

[patch] autodomain (SIP Domain Support): Add only really different domain
with TLS.
(Reported by Alexander Traud)

   - [ASTERISK-27100
   <https://issues.asterisk.org/jira/browse/ASTERISK-27100>] -

channel: ast_waitfordigit_full fails to clear flag in an error branch.
(Reported by Corey Farrell)

   - [ASTERISK-27090
   <https://issues.asterisk.org/jira/browse/ASTERISK-27090>] -

PJSIP: Deadlock using TCP transport
(Reported by Richard Mudgett)

   - [ASTERISK-25665
   <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] -

Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)

   - [ASTERISK-27065
   <https://issues.asterisk.org/jira/browse/ASTERISK-27065>] -

call hangup after leaving app_queue
(Reported by Marek Cervenka)

   - [ASTERISK-26978
   <https://issues.asterisk.org/jira/browse/ASTERISK-26978>] -

rtp: Crash in ast_rtp_codecs_payload_code()
(Reported by Ross Beer)

   - [ASTERISK-24052
   <https://issues.asterisk.org/jira/browse/ASTERISK-24052>] -

app_voicemail reloads result in leaked IMAP sockets.
(Reported by Louis Jocelyn Paquet)

   - [ASTERISK-27074
   <https://issues.asterisk.org/jira/browse/ASTERISK-27074>] -

core_local: local channel data not being properly unref'ed and unlocked
(Reported by Kevin Harwell)

   - [ASTERISK-27075
   <https://issues.asterisk.org/jira/browse/ASTERISK-27075>] -

bridge: stuck channel(s) after failed attended transfer
(Reported by Kevin Harwell)

   - [ASTERISK-27060
   <https://issues.asterisk.org/jira/browse/ASTERISK-27060>] -

Comment typo format_g729.c
(Reported by Matthew Fredrickson)

   - [ASTERISK-27041
   <https://issues.asterisk.org/jira/browse/ASTERISK-27041>] -

Core/PBX: [patch] Deadlock between dialplan execution and application
unregistration
(Reported by Frederic LE FOLL)

   - [ASTERISK-27026
   <https://issues.asterisk.org/jira/browse/ASTERISK-27026>] -

res_ari: Crash when no ari.conf configuration file exists
(Reported by Ronald Raikes)

   - [ASTERISK-27057
   <https://issues.asterisk.org/jira/browse/ASTERISK-27057>] -

Seg Fault in ast_sorcery_object_get_id at sorcery.c
(Reported by Ryan Smith)

   - [ASTERISK-27024
   <https://issues.asterisk.org/jira/browse/ASTERISK-27024>] -

nat/external_media settings ignored in 14.4.1
(Reported by Christopher van de Sande)

   - [ASTERISK-27046
   <https://issues.asterisk.org/jira/browse/ASTERISK-27046>] -

res_pjsip_transport_websocket: segfault in get_write_timeout
(Reported by Jørgen H)

   - [ASTERISK-27022
   <https://issues.asterisk.org/jira/browse/ASTERISK-27022>] -

res_rtp_asterisk: Incorrect SSRC change for RTCP component
(Reported by Michael Walton)

   - [ASTERISK-26923
   <https://issues.asterisk.org/jira/browse/ASTERISK-26923>] -

bridging: T.38 request is lost when channels are added to bridge
(Reported by Torrey Searle)

   - [ASTERISK-27053
   <https://issues.asterisk.org/jira/browse/ASTERISK-27053>] -

res_pjsip_refer/session: Calls dropped during transfer
(Reported by Kevin Harwell)

   - [ASTERISK-27052
   <https://issues.asterisk.org/jira/browse/ASTERISK-27052>] -

Asterisk build process fails with flag --with-pjproject-bundled with curl
download command and slow network
(Reported by alex)

   - [ASTERISK-27039
   <https://issues.asterisk.org/jira/browse/ASTERISK-27039>] -

chan_pjsip: Device state is idle when channel from endpoint is in early
media
(Reported by Joshua Colp)

   - [ASTERISK-26996
   <https://issues.asterisk.org/jira/browse/ASTERISK-26996>] -

chan_pjsip: Flipping between codecs
(Reported by Michael Maier)

   - [ASTERISK-26281
   <https://issues.asterisk.org/jira/browse/ASTERISK-26281>] -

chan_pjsip would send INVITE to 'Unreachable' endpoints
(Reported by Jacek Konieczny)

   - [ASTERISK-26973
   <https://issues.asterisk.org/jira/browse/ASTERISK-26973>] -

bridge: Crash when freeing frame and snooping
(Reported by Michel R. Vaillancourt)

   - [ASTERISK-19291
   <https://issues.asterisk.org/jira/browse/ASTERISK-19291>] -

Background in realtime
(Reported by Andrew Nowrot)

   - [ASTERISK-27025
   <https://issues.asterisk.org/jira/browse/ASTERISK-27025>] -

channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)

   - [ASTERISK-27021
   <https://issues.asterisk.org/jira/browse/ASTERISK-27021>] -

GET /recordings/stored returns 500 Internal Server Error
(Reported by Tim Morgan)

   - [ASTERISK-24858
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-23951>] -

Asterisk attempts and fails to build format_mp3 even if mp3lib was not
downloaded
(Reported by Tzafrir Cohen)

   - [ASTERISK-25294
   <https://issues.asterisk.org/jira/browse/ASTERISK-25294>] -

srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)

   - [ASTERISK-23839
   <https://issues.asterisk.org/jira/browse/ASTERISK-23839>] -

AGI - RECORD FILE - documentation doesn't describe BEEP argument
(Reported by Rusty Newton)

   - [ASTERISK-22432
   <https://issues.asterisk.org/jira/browse/ASTERISK-22432>] -

Async AGI crashes Asterisk when issuing "set variable" command without args
(Reported by Antoine Pitrou)

   - [ASTERISK-25662
   <https://issues.asterisk.org/jira/browse/ASTERISK-25662>] -

Malformed AGI 520 Usage response
(Reported by Tony Mountifield)

   - [ASTERISK-27008
   <https://issues.asterisk.org/jira/browse/ASTERISK-27008>] -

res_format_attr_h264: SDP parse fails if fmtp optional parameters have a
space
(Reported by John Harris)

   - [ASTERISK-26399
   <https://issues.asterisk.org/jira/browse/ASTERISK-26399>] -

app_queue: Agent not called when caller is parked
(Reported by wushumasters)

   - [ASTERISK-26400
   <https://issues.asterisk.org/jira/browse/ASTERISK-26400>] -

app_queue: Queue member stops being called after AMI "Redirect" action for
queues with wrapuptime
(Reported by Etienne Lessard)

   - [ASTERISK-26715
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-27012>] -

app_confbridge: ConfBridge sometimes does not play user name recording
while leaving
(Reported by Robert Mordec)

   - [ASTERISK-26979
   <https://issues.asterisk.org/jira/browse/ASTERISK-26979>] -

res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or
110
(Reported by Javier Riveros )

   - [ASTERISK-26982
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-26964>] -

res_pjsip_session: Wrong From on reinvite when request and To URI differ
(Reported by Yasin CANER)

   - [ASTERISK-26789
   <https://issues.asterisk.org/jira/browse/ASTERISK-26789>] -

Audit manipulation of channel flags without locks
(Reported by Joshua Colp)

   - [ASTERISK-26333
   <https://issues.asterisk.org/jira/browse/ASTERISK-26333>] -

Problems with Blind Transfer, PJSIP (Aastra 6869i)
(Reported by Matthias Binder)

*Improvements made in this release:*
-----------------------------------

   - [ASTERISK-26230
   <https://issues.asterisk.org/jira/browse/ASTERISK-26230>] -

[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on
startup
(Reported by Alexei Gradinari)

   - [ASTERISK-27043
   <https://issues.asterisk.org/jira/browse/ASTERISK-27043>] -

Core/BuildSystem: Add defines to fix build with LibreSSL
(Reported by Guido Falsi)

   - [ASTERISK-27042
   <https://issues.asterisk.org/jira/browse/ASTERISK-27042>] -

Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h
file
(Reported by Guido Falsi)

   - [ASTERISK-26419
   <https://issues.asterisk.org/jira/browse/ASTERISK-26419>] -

audiohooks: Remove redundant codec translations when using audiohooks
(Reported by Michael Walton)

   - [ASTERISK-26976
   <https://issues.asterisk.org/jira/browse/ASTERISK-26976>] -

libsrtp-2.x.x support
(Reported by Alex)

   - [ASTERISK-26124
   <https://issues.asterisk.org/jira/browse/ASTERISK-26124>] -

res_agi: Set audio format for EAGI audio stream
(Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0

*Thank you for your continued support of Asterisk!*

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