Rilasciato Asterisk 13.16.0-rc1

Il giorno 22 maggio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.16.0-rc1.

Dal post originale:

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

Improvements made in this release:

[ASTERISK-26088] -
Investigate heavy memory utilization by res_pjsip_pubsub
(Reported by Richard Mudgett)
[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 candidate, please see the ChangeLog:

Commenti (0) Trackback (0)

Spiacenti, il modulo dei commenti è chiuso per ora.

Ancora nessun trackback.