[AsteriskBrasil] [asterisk-dev] Asterisk 13.16.0 Now Available
Sylvio Jollenbeck
sylvio.jollenbeck em gmail.com
Terça Maio 30 16:55:51 BRT 2017
The Asterisk Development Team would like to announce the release of
Asterisk 13.16.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.16.0 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-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-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-25665
<https://issues.asterisk.org/jira/browse/ASTERISK-25665>] -
Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
- [ASTERISK-26998
<https://issues.asterisk.org/jira/browse/ASTERISK-26998>] -
res_pjsip_session: INVITE retransmissions could still setup the same call
again.
(Reported by Richard Mudgett)
- [ASTERISK-26143
<https://issues.asterisk.org/jira/browse/ASTERISK-26143>] -
res_rtp_asterisk: One way audio when transcoding
(Reported by Henning Holtschneider)
- [ASTERISK-26606
<https://issues.asterisk.org/jira/browse/ASTERISK-26606>] -
tcptls: Incorrect OpenSSL function call leads to misleading error report
(Reported by Bob Ham)
- [ASTERISK-26983
<https://issues.asterisk.org/jira/browse/ASTERISK-26983>] -
Crash in Manager Reload when TLS Config Changes
(Reported by Joshua Elson)
- [ASTERISK-25032
<https://issues.asterisk.org/jira/browse/ASTERISK-25032>] -
[patch]cel_odbc sometimes inserts CEL with wrong eventtime
(Reported by Etienne Lessard)
- [ASTERISK-26173
<https://issues.asterisk.org/jira/browse/ASTERISK-26173>] -
func_cdr: CDR function does not permit empty values to be assigned
(Reported by gkloepfer)
- [ASTERISK-25506
<https://issues.asterisk.org/jira/browse/ASTERISK-25506>] -
[patch]CONFBRIDGE failure after an app_confbrige.so module reload results
in segfault or error/warning messages.
(Reported by Frederic LE FOLL)
- [ASTERISK-24529
<https://issues.asterisk.org/jira/browse/ASTERISK-24529>] -
Using AMI Action Bridge to on an already bridged channel causes the
incorrect return priority to be used
(Reported by Corey Farrell)
- [ASTERISK-26860
<https://issues.asterisk.org/jira/browse/ASTERISK-26860>] -
Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port
missing in (null)
(Reported by Evers Lab)
- [ASTERISK-26922
<https://issues.asterisk.org/jira/browse/ASTERISK-26922>] -
chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
- [ASTERISK-26974
<https://issues.asterisk.org/jira/browse/ASTERISK-26974>] -
res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
- [ASTERISK-26908
<https://issues.asterisk.org/jira/browse/ASTERISK-26908>] -
res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
(Reported by Richard Mudgett)
- [ASTERISK-25823
<https://issues.asterisk.org/jira/browse/ASTERISK-25823>] -
SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or
directory.
(Reported by Andreas Krüger)
- [ASTERISK-26951
<https://issues.asterisk.org/jira/browse/ASTERISK-26951>] -
chan_sip: ACK with SDP does not update a direct media bridge
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26930
<https://issues.asterisk.org/jira/browse/ASTERISK-26930>] -
pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2
instrunction Linux
(Reported by abelbeck)
- [ASTERISK-26926
<https://issues.asterisk.org/jira/browse/ASTERISK-26926>] -
func_speex: Crash caused by frame with no datalen
(Reported by Richard Kenner)
- [ASTERISK-26929
<https://issues.asterisk.org/jira/browse/ASTERISK-26929>] -
pjsip: Add database tables for RLS
(Reported by Joshua Colp)
- [ASTERISK-26953
<https://issues.asterisk.org/jira/browse/ASTERISK-26953>] -
Asterisk crash if hep.conf have some missing parameters
(Reported by Joel Vandal)
- [ASTERISK-26890
<https://issues.asterisk.org/jira/browse/ASTERISK-26890>] -
STUN server with non-default-route transport causes INVITE delay
(Reported by George Joseph)
- [ASTERISK-26692
<https://issues.asterisk.org/jira/browse/ASTERISK-26692>] -
res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk
(using chan_sip)
(Reported by scgm11)
- [ASTERISK-26835
<https://issues.asterisk.org/jira/browse/ASTERISK-26835>] -
res_rtp_asterisk: Crash when freeing RTCP address string
(Reported by Niklas Larsson)
- [ASTERISK-26853
<https://issues.asterisk.org/jira/browse/ASTERISK-26853>] -
res_rtp_asterisk: Crash in pjnath when receiving packet
(Reported by Adagio)
- [ASTERISK-26613
<https://issues.asterisk.org/jira/browse/ASTERISK-26613>] -
format_wav: wav16 format read file only by 320 - half of frame
(Reported by Vitaly K)
- [ASTERISK-26169
<https://issues.asterisk.org/jira/browse/ASTERISK-26169>] -
format_ogg_vorbis: Memory leak using OGG in MixMonitor
(Reported by Ivan Myalkin)
- [ASTERISK-21856
<https://issues.asterisk.org/jira/browse/ASTERISK-21856>] -
STUN never works when asterisk started without internet access
(Reported by Jeremy Kister)
- [ASTERISK-20984
<https://issues.asterisk.org/jira/browse/ASTERISK-20984>] -
Audible clicks when playing sox encoded au file with STREAM FILE AGI command
(Reported by Roman S.)
- [ASTERISK-26851
<https://issues.asterisk.org/jira/browse/ASTERISK-26851>] -
res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26903
<https://issues.asterisk.org/jira/browse/ASTERISK-26903>] -
Listening TCP/TLS sockets stop when temporarily out of open files
(Reported by Walter Doekes)
- [ASTERISK-26528
<https://issues.asterisk.org/jira/browse/ASTERISK-26528>] -
[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
(Reported by Badalian Vyacheslav)
- [ASTERISK-26928
<https://issues.asterisk.org/jira/browse/ASTERISK-26928>] -
pjsip: Add database tables for PUBLISH support
(Reported by Joshua Colp)
- [ASTERISK-26927
<https://issues.asterisk.org/jira/browse/ASTERISK-26927>] -
pjproject_bundled: Crash on pj_ssl_get_info() while
ioqueue_on_read_complete().
(Reported by Alexander Traud)
- [ASTERISK-26905
<https://issues.asterisk.org/jira/browse/ASTERISK-26905>] -
pjproject_bundled: Merge 3 upstream deadlock patches into bundled
(Reported by Ross Beer)
- [ASTERISK-26897
<https://issues.asterisk.org/jira/browse/ASTERISK-26897>] -
chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-25974
<https://issues.asterisk.org/jira/browse/ASTERISK-25974>] -
Unused realtime MOH classes not purged on 'moh reload'
(Reported by Sébastien Couture)
- [ASTERISK-26916
<https://issues.asterisk.org/jira/browse/ASTERISK-26916>] -
res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-21721
<https://issues.asterisk.org/jira/browse/ASTERISK-21721>] -
SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
- [ASTERISK-26915
<https://issues.asterisk.org/jira/browse/ASTERISK-26915>] -
chan_sip: Session Timers required but refused wrongly.
(Reported by Alexander Traud)
- [ASTERISK-26363
<https://issues.asterisk.org/jira/browse/ASTERISK-26363>] -
res_pjsip: Bye sent to sip trunk is not authenticated even after receiving
a 407 error code
(Reported by Yaacov Akiba Slama)
- [ASTERISK-26896
<https://issues.asterisk.org/jira/browse/ASTERISK-26896>] -
Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
(Reported by twisted)
- [ASTERISK-26705
<https://issues.asterisk.org/jira/browse/ASTERISK-26705>] -
libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-21009
<https://issues.asterisk.org/jira/browse/ASTERISK-21009>] -
xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub
unsubscription on client
(Reported by Marcello Ceschia)
- [ASTERISK-25490
<https://issues.asterisk.org/jira/browse/ASTERISK-25490>] -
[patch]SDP crypto tag is validated incorrectly
(Reported by Joerg Sonnenberger)
- [ASTERISK-24712
<https://issues.asterisk.org/jira/browse/ASTERISK-24712>] -
xmpp: starttls problem causes connection spew
(Reported by Matthias Urlichs)
- [ASTERISK-26086
<https://issues.asterisk.org/jira/browse/ASTERISK-26086>] -
res_musiconhold: format option is not documented adequately
(Reported by Jens Bürger)
- [ASTERISK-23996
<https://issues.asterisk.org/jira/browse/ASTERISK-23996>] -
No core dumps because of res_musiconhold chdir.
(Reported by Walter Doekes)
- [ASTERISK-26814
<https://issues.asterisk.org/jira/browse/ASTERISK-26814>] -
pjproject_bundled build fails to download pjproject source when using cURL
(Reported by Gergely Dömsödi)
- [ASTERISK-23510
<https://issues.asterisk.org/jira/browse/ASTERISK-23510>] -
JABBER_STATUS fails with improper code 7 for unavailable clients
(Reported by Anthony Critelli)
- [ASTERISK-21855
<https://issues.asterisk.org/jira/browse/ASTERISK-21855>] -
Asterisk crashes when XMPP message is sent (JabberSend) and no internet
connection is available
(Reported by Jeremy Kister)
- [ASTERISK-25622
<https://issues.asterisk.org/jira/browse/ASTERISK-25622>] -
WARNING for "JABBER: socket read error" should be more specific
(Reported by Sean Darcy)
- [ASTERISK-26818
<https://issues.asterisk.org/jira/browse/ASTERISK-26818>] -
cdr: Problem setting variables in h exten
(Reported by scgm11)
- [ASTERISK-26875
<https://issues.asterisk.org/jira/browse/ASTERISK-26875>] -
app_mixmonitor: Recording out of sync when 183 but no RTP
(Reported by Aaron An)
*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26088
<https://issues.asterisk.org/jira/browse/ASTERISK-26088>] -
Investigate heavy memory utilization by res_pjsip_pubsub
(Reported by Richard Mudgett)
- [ASTERISK-26427
<https://issues.asterisk.org/jira/browse/ASTERISK-26427>] -
res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp
when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0
*Thank you for your continued support of Asterisk!*
--
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