[AsteriskBrasil] [asterisk-dev] Asterisk 13.19.0 Now Available
Sylvio Jollenbeck
sylvio.jollenbeck em gmail.com
Quinta Janeiro 11 20:45:16 BRST 2018
The Asterisk Development Team would like to announce the release of
Asterisk 13.19.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.19.0 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:
*New Features made in this release:*
-----------------------------------
- [ASTERISK-27478
<https://issues.asterisk.org/jira/browse/ASTERISK-27478>] -
PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI.
(Reported by Richard Mudgett)
- [ASTERISK-27413
<https://issues.asterisk.org/jira/browse/ASTERISK-27413>] -
Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
- [ASTERISK-27206
<https://issues.asterisk.org/jira/browse/ASTERISK-27206>] -
res_pjsip: No mechanism exists to limit endpoint identification to IP only
(Reported by Ben Merrills)
*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-27531
<https://issues.asterisk.org/jira/browse/ASTERISK-27531>] -
Compiler optimizations can break module load sequence.
(Reported by abelbeck)
- [ASTERISK-27480
<https://issues.asterisk.org/jira/browse/ASTERISK-27480>] -
Security: Authenticated SUBSCRIBE without Contact crashes asterisk
(Reported by Ross Beer)
- [ASTERISK-27299
<https://issues.asterisk.org/jira/browse/ASTERISK-27299>] -
Asterisk Hangs with Bad file descriptor on read()
(Reported by Abhay Gupta)
- [ASTERISK-25079
<https://issues.asterisk.org/jira/browse/ASTERISK-25079>] -
AMI bridge of channels results in MOH not destroyed and robotic audio on
one channel
(Reported by Zane Conkle)
- [ASTERISK-27490
<https://issues.asterisk.org/jira/browse/ASTERISK-27490>] -
chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
- [ASTERISK-24756
<https://issues.asterisk.org/jira/browse/ASTERISK-24756>] -
ConfBridge sound_muted does not work from CLI or AMI
(Reported by Thomas Frederiksen)
- [ASTERISK-25649
<https://issues.asterisk.org/jira/browse/ASTERISK-25649>] -
Transfer application does not work with Local channels - documentation
misleading
(Reported by Ivan Ullmann)
- [ASTERISK-25869
<https://issues.asterisk.org/jira/browse/ASTERISK-25869>] -
chan_sip: "rejected because extension not found" should be logged as a
security event
(Reported by Brian J. Murrell)
- [ASTERISK-27440
<https://issues.asterisk.org/jira/browse/ASTERISK-27440>] -
Strictrtp has issues to qualify video rtp streams
(Reported by Wim De Vlaminck)
- [ASTERISK-24329
<https://issues.asterisk.org/jira/browse/ASTERISK-24329>] -
Music On Hold announcement cuts intro of music the first time it is played
(Reported by Thomas Frederiksen)
- [ASTERISK-19657
<https://issues.asterisk.org/jira/browse/ASTERISK-19657>] -
Coverity Report: Fix issues for error type CHAR_IO
(Reported by Matt Jordan)
- [ASTERISK-27175
<https://issues.asterisk.org/jira/browse/ASTERISK-27175>] -
iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
- [ASTERISK-27430
<https://issues.asterisk.org/jira/browse/ASTERISK-27430>] -
README refers to security documents that do not exist.
(Reported by Corey Farrell)
- [ASTERISK-20281
<https://issues.asterisk.org/jira/browse/ASTERISK-20281>] -
"core set verbose" behaves strangely, can't alias it, cli.conf example
broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
- [ASTERISK-27382
<https://issues.asterisk.org/jira/browse/ASTERISK-27382>] -
crash after an invalid rtcp packet from GT48 FXS gateway
(Reported by Tzafrir Cohen)
- [ASTERISK-27429
<https://issues.asterisk.org/jira/browse/ASTERISK-27429>] -
res_rtp_asterisk: Multiple reports in an RTCP packet will write past where
it should
(Reported by Vitezslav Novy)
- [ASTERISK-27408
<https://issues.asterisk.org/jira/browse/ASTERISK-27408>] -
Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
- [ASTERISK-18411
<https://issues.asterisk.org/jira/browse/ASTERISK-18411>] -
Queue members with hints for state_interface get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
- [ASTERISK-26131
<https://issues.asterisk.org/jira/browse/ASTERISK-26131>] -
chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a
call to a single character in a dot pattern match
(Reported by Dwayne Hubbard)
- [ASTERISK-27475
<https://issues.asterisk.org/jira/browse/ASTERISK-27475>] -
codec_opus requires libcurl
(Reported by Samuel For)
- [ASTERISK-27467
<https://issues.asterisk.org/jira/browse/ASTERISK-27467>] -
pjsip_options: qualify_frequency sometimes not applied on reload
(Reported by John Bigelow)
- [ASTERISK-27465
<https://issues.asterisk.org/jira/browse/ASTERISK-27465>] -
CLI Completion Not Working
(Reported by Ross Beer)
- [ASTERISK-27460
<https://issues.asterisk.org/jira/browse/ASTERISK-27460>] -
CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
- [ASTERISK-27453
<https://issues.asterisk.org/jira/browse/ASTERISK-27453>] -
RTP: Blind transfer direct media scenario results in one way audio.
(Reported by Richard Mudgett)
- [ASTERISK-20643
<https://issues.asterisk.org/jira/browse/ASTERISK-20643>] -
SIP ICE support - remove hardcoded limitation on SDP size, make ICE support
disabled by default in SIP, maybe provide a better warning message
(Reported by Roy)
- [ASTERISK-26980
<https://issues.asterisk.org/jira/browse/ASTERISK-26980>] -
pjsip: Clean up WebRTC disables
(Reported by abelbeck)
- [ASTERISK-27452
<https://issues.asterisk.org/jira/browse/ASTERISK-27452>] -
Security: chan_skinny: Memory exhaustion if flooded with unauthenticated
requests
(Reported by George Joseph)
- [ASTERISK-27454
<https://issues.asterisk.org/jira/browse/ASTERISK-27454>] -
res_http_post: Don't require GMIME_MAJOR_VERSION
(Reported by Joshua Colp)
- [ASTERISK-23735
<https://issues.asterisk.org/jira/browse/ASTERISK-23735>] -
Transcoding makes bad choice in high-rate translations
(Reported by Richard Kenner)
- [ASTERISK-27445
<https://issues.asterisk.org/jira/browse/ASTERISK-27445>] -
ARI: Updating a bridge gives wrong error message.
(Reported by Frank Durden)
- [ASTERISK-24662
<https://issues.asterisk.org/jira/browse/ASTERISK-24662>] -
[patch] column and row headers for Signed Linear format variants in output
of 'core show translation' are ambiguous
(Reported by Rusty Newton)
- [ASTERISK-27353
<https://issues.asterisk.org/jira/browse/ASTERISK-27353>] -
H323 audio starts with a delay of 2 seconds.
(Reported by Marco Giordani)
- [ASTERISK-27442
<https://issues.asterisk.org/jira/browse/ASTERISK-27442>] -
pjsip: 183 without To tag does not negotiate media
(Reported by Kevin Harwell)
- [ASTERISK-27437
<https://issues.asterisk.org/jira/browse/ASTERISK-27437>] -
[patch] ICE: server-reflexive candidates (srflx) with Dual-Stack.
(Reported by Alexander Traud)
- [ASTERISK-27434
<https://issues.asterisk.org/jira/browse/ASTERISK-27434>] -
[patch] chan_sip/ICE: Square brackets around IPv6 addresses.
(Reported by Alexander Traud)
- [ASTERISK-27435
<https://issues.asterisk.org/jira/browse/ASTERISK-27435>] -
[patch] configure: pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
- [ASTERISK-27431
<https://issues.asterisk.org/jira/browse/ASTERISK-27431>] -
Asterisk fails to build when openssl headers are not installed.
(Reported by Corey Farrell)
- [ASTERISK-27332
<https://issues.asterisk.org/jira/browse/ASTERISK-27332>] -
Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
- [ASTERISK-27421
<https://issues.asterisk.org/jira/browse/ASTERISK-27421>] -
RTP source learning not working with devices that have some clock issues
(Reported by nappsoft)
- [ASTERISK-27361
<https://issues.asterisk.org/jira/browse/ASTERISK-27361>] -
Attended transfer crashes in Asterisk 13.17.2
(Reported by Alessandro Pimenta)
- [ASTERISK-27238
<https://issues.asterisk.org/jira/browse/ASTERISK-27238>] -
Bridging: Crash freeing a frame that's already been freed
(Reported by Richard Kenner)
- [ASTERISK-27412
<https://issues.asterisk.org/jira/browse/ASTERISK-27412>] -
core: Audiohook freeing interpolated frame when it shouldn't.
(Reported by Mikhail)
- [ASTERISK-27423
<https://issues.asterisk.org/jira/browse/ASTERISK-27423>] -
app_record: We set the RECORD_STATUS channel variable before closing the
file
(Reported by George Joseph)
- [ASTERISK-26758
<https://issues.asterisk.org/jira/browse/ASTERISK-26758>] -
res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in
"source ip address" and "destination ip address" fields in HEP packets
(Reported by Max Norba)
- [ASTERISK-27363
<https://issues.asterisk.org/jira/browse/ASTERISK-27363>] -
res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress)
(Reported by Vasilii Rogin)
- [ASTERISK-27415
<https://issues.asterisk.org/jira/browse/ASTERISK-27415>] -
asterisk.conf: Setting astctl without setting astrundir is ineffective.
(Reported by Corey Farrell)
- [ASTERISK-27411
<https://issues.asterisk.org/jira/browse/ASTERISK-27411>] -
pjsip: TCP connections may not be destroyed
(Reported by Joshua Colp)
- [ASTERISK-27345
<https://issues.asterisk.org/jira/browse/ASTERISK-27345>] -
res_pjsip_session: RTP instances leak on 488 responses.
(Reported by Corey Farrell)
- [ASTERISK-27337
<https://issues.asterisk.org/jira/browse/ASTERISK-27337>] -
chan_sip: Security vulnerability with client code header (revisited)
(Reported by Richard Mudgett)
- [ASTERISK-27319
<https://issues.asterisk.org/jira/browse/ASTERISK-27319>] -
(Security) Function in PJSIP 2.7 miscalculates the length of an unsigned
long variable in 64bit machines
(Reported by Kim youngsung)
- [ASTERISK-27391
<https://issues.asterisk.org/jira/browse/ASTERISK-27391>] -
Regression: Deadlock between AOR named lock and pjproject grp lock
(Reported by shaurya jain)
- [ASTERISK-27393
<https://issues.asterisk.org/jira/browse/ASTERISK-27393>] -
res_pjsip: Crash occurs when an empty contact read from astdb or database
(Reported by Aaron An)
- [ASTERISK-27290
<https://issues.asterisk.org/jira/browse/ASTERISK-27290>] -
res_pjsip: PIDF contact field has malformed/invalid XML
(Reported by basildane)
- [ASTERISK-27032
<https://issues.asterisk.org/jira/browse/ASTERISK-27032>] -
res_pjsip: TLS options do not handle empty values
(Reported by seanchann.zhou)
- [ASTERISK-27394
<https://issues.asterisk.org/jira/browse/ASTERISK-27394>] -
[patch] tcptls: Print notice when TLS is enabled but not configured.
(Reported by Alexander Traud)
- [ASTERISK-26426
<https://issues.asterisk.org/jira/browse/ASTERISK-26426>] -
format_ogg_opus: remove from source
(Reported by Kevin Harwell)
- [ASTERISK-27378
<https://issues.asterisk.org/jira/browse/ASTERISK-27378>] -
Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
- [ASTERISK-27387
<https://issues.asterisk.org/jira/browse/ASTERISK-27387>] -
Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more
(Reported by Michael Maier)
- [ASTERISK-27390
<https://issues.asterisk.org/jira/browse/ASTERISK-27390>] -
Audit menuselect module dependencies
(Reported by Corey Farrell)
- [ASTERISK-27389
<https://issues.asterisk.org/jira/browse/ASTERISK-27389>] -
Optional API modules should not allow unload.
(Reported by Corey Farrell)
- [ASTERISK-27369
<https://issues.asterisk.org/jira/browse/ASTERISK-27369>] -
Bridge() dialplan application fails without setting BRIDGERESULT channel
variable
(Reported by James Terhune)
- [ASTERISK-27377
<https://issues.asterisk.org/jira/browse/ASTERISK-27377>] -
Typo in CHANNEL(dtmf_features) usage documentation
(Reported by Igor Goncharovsky)
- [ASTERISK-27181
<https://issues.asterisk.org/jira/browse/ASTERISK-27181>] -
GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting'
(Reported by Anthony Messina)
- [ASTERISK-27194
<https://issues.asterisk.org/jira/browse/ASTERISK-27194>] -
jitterbuffer: Does not handle case where translator returns null frame.
(Reported by Joshua Elson)
- [ASTERISK-26639
<https://issues.asterisk.org/jira/browse/ASTERISK-26639>] -
core: Disabling xmldoc support does not work. Also results in abort during
Asterisk startup.
(Reported by Mr Dini)
- [ASTERISK-27372
<https://issues.asterisk.org/jira/browse/ASTERISK-27372>] -
ARI: Node ARI client broken in latest versions of 13 and 14
(Reported by Benjamin Keith Ford)
- [ASTERISK-18140
<https://issues.asterisk.org/jira/browse/ASTERISK-18140>] -
Expires handling in SUBSCRIBE confuses the absence of the Expires header
field with an unsubscribe action.
(Reported by Jonathan Cloots)
- [ASTERISK-25960
<https://issues.asterisk.org/jira/browse/ASTERISK-25960>] -
The config_hook unit test causes Asterisk to crash if run a second time
(Reported by George Joseph)
- [ASTERISK-27198
<https://issues.asterisk.org/jira/browse/ASTERISK-27198>] -
res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
- [ASTERISK-27346
<https://issues.asterisk.org/jira/browse/ASTERISK-27346>] -
res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
(Reported by Ronald Raikes)
- [ASTERISK-27365
<https://issues.asterisk.org/jira/browse/ASTERISK-27365>] -
[patch] chan_sip: Crypto attribute not last but first on SDP media level.
(Reported by Alexander Traud)
- [ASTERISK-24483
<https://issues.asterisk.org/jira/browse/ASTERISK-24483>] -
res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id ==
-1
(Reported by Tzafrir Cohen)
- [ASTERISK-23462
<https://issues.asterisk.org/jira/browse/ASTERISK-23462>] -
Cannot disable SIP debugging via CLI after enabling with conf file option -
also 'sip set debug off' reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
- [ASTERISK-27328
<https://issues.asterisk.org/jira/browse/ASTERISK-27328>] -
Missing openssl dependencies in res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
- [ASTERISK-27341
<https://issues.asterisk.org/jira/browse/ASTERISK-27341>] -
[patch] res_pjsip_session: SIP/SDP origin (o=) contains local address.
(Reported by Alexander Traud)
- [ASTERISK-27343
<https://issues.asterisk.org/jira/browse/ASTERISK-27343>] -
Fails to build in FreeBSD due to sys/sysmacros.h not existing there
(Reported by Guido Falsi)
- [ASTERISK-27340
<https://issues.asterisk.org/jira/browse/ASTERISK-27340>] -
backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
- [ASTERISK-27339
<https://issues.asterisk.org/jira/browse/ASTERISK-27339>] -
[patch] Crash on ast_ssl_teardown when stopping.
(Reported by Alexander Traud)
- [ASTERISK-27333
<https://issues.asterisk.org/jira/browse/ASTERISK-27333>] -
sip_to_pjsip not correctly handling disallow=all directive
(Reported by Torrey Searle)
*Improvements made in this release:*
-----------------------------------
- [ASTERISK-24297
<https://issues.asterisk.org/jira/browse/ASTERISK-24297>] -
cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
- [ASTERISK-27449
<https://issues.asterisk.org/jira/browse/ASTERISK-27449>] -
[PATCH] When failing to acquire target during attended transfer, display
wanted extension
(Reported by Niklas Larsson)
- [ASTERISK-27456
<https://issues.asterisk.org/jira/browse/ASTERISK-27456>] -
app_voicemail: Add new object for VoicemailUserEntry
(Reported by sungtae kim)
- [ASTERISK-27380
<https://issues.asterisk.org/jira/browse/ASTERISK-27380>] -
ast_coredumper: allow pointing out the asterisk binary explicitly
(Reported by Tzafrir Cohen)
- [ASTERISK-23556
<https://issues.asterisk.org/jira/browse/ASTERISK-23556>] -
Compilation warning for invert.c (array subscript is above array bounds)
(Reported by Marcello Ceschia)
- [ASTERISK-27355
<https://issues.asterisk.org/jira/browse/ASTERISK-27355>] -
Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
- [ASTERISK-27335
<https://issues.asterisk.org/jira/browse/ASTERISK-27335>] -
CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
*Thank you for your continued support of Asterisk!*
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