[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 13.17.0 Now Available
Sylvio Jollenbeck
sylvio.jollenbeck em gmail.com
Quarta Julho 12 11:19:26 BRT 2017
The Asterisk Development Team would like to announce the release of
Asterisk 13.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.17.0 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-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-24052
<https://issues.asterisk.org/jira/browse/ASTERISK-24052>] -
app_voicemail reloads result in leaked IMAP sockets.
(Reported by Louis Jocelyn Paquet)
- [ASTERISK-27060
<https://issues.asterisk.org/jira/browse/ASTERISK-27060>] -
Comment typo format_g729.c
(Reported by Matthew Fredrickson)
- [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-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-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-25101
<https://issues.asterisk.org/jira/browse/ASTERISK-25101>] -
DTLS configuration can not be specified in the general section -
documentation
(Reported by Ben Langfeld)
- [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-13.17.0
*Thank you for your continued support of Asterisk!*
--
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