[asterisk-dev] Asterisk 13.1.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Mon Dec 8 14:01:38 CST 2014
The Asterisk Development Team has announced the first release candidate of
Asterisk 13.1.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release candidate:
New Features made in this release:
-----------------------------------
* ASTERISK-24554 - AMI/ARI: Generate events on connected line
changes (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
Corey Farrell)
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
leak (Reported by Corey Farrell)
* ASTERISK-24430 - missing letter "p" in word response in
OriginateResponse event documentation (Reported by Dafi Ni)
* ASTERISK-24437 - Review implementation of ast_bridge_impart for
leaks and document proper usage (Reported by Scott Griepentrog)
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
Corey Farrell)
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
Corey Farrell)
* ASTERISK-24458 - chan_phone fails to build on big endian systems
(Reported by Tzafrir Cohen)
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
* ASTERISK-24304 - asterisk crashing randomly because of unistim
channel (Reported by dhanapathy sathya)
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
Nick Adams)
* ASTERISK-24462 - res_pjsip: Stale qualify statistics after
disablementation (Reported by Kevin Harwell)
* ASTERISK-24465 - audiohooks list leaks reference to formats
(Reported by Corey Farrell)
* ASTERISK-24466 - app_queue: fix a couple leaks to struct
call_queue (Reported by Corey Farrell)
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
(Reported by Corey Farrell)
* ASTERISK-24411 - [patch] Status of outbound registration is not
changed upon unregistering. (Reported by John Bigelow)
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
leaks (Reported by Corey Farrell)
* ASTERISK-24480 - res_http_websockets: Module reference decrease
below zero (Reported by Corey Farrell)
* ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
audiohook callback (Reported by Corey Farrell)
* ASTERISK-24487 - configuration: sections should be loadable as
template even when not marked (Reported by Scott Griepentrog)
* ASTERISK-20127 - [Regression] Config.c config_text_file_load()
unescapes semicolons ("\;" -> ";") turning them into comments
(corruption) on rewrite of a config file (Reported by George
Joseph)
* ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
when DNS settings invalid (Reported by Melissa Shepherd)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
* ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
Conkle)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
extra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
waiting for more matching digits. (Reported by Richard Mudgett)
* ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
queue caller (Reported by Steve Pitts)
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)
* ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
header fix (Reported by abelbeck)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
length exceeds 50 (roughly) national symbols (Reported by
Dmitriy Bubnov)
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
revision r227276 (Reported by Xavier Hienne)
* ASTERISK-24505 - manager: http connections leak references
(Reported by Corey Farrell)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
coverage are enabled (Reported by Corey Farrell)
* ASTERISK-24444 - PBX: Crash when generating extension for
pattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
packet to JSON for res_hep_rtcp and report blocks are greater
than 1 (Reported by Gregory Malsack)
* ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
transfer (Reported by Beppo Mazzucato)
* ASTERISK-24501 - ARI: Moving a channel between bridges followed
by a hangup can cause an ARI client to not receive an expected
ChannelLeftBridge event before StasisEnd (Reported by Matt
Jordan)
* ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
(Reported by Leon Rowland)
* ASTERISK-23651 - Reloading some modules that are loaded already,
results in 'No such module' before a successful reload (Reported
by Rusty Newton)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
endmarked users when last marked user leaves (Reported by Matt
Jordan)
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
(Reported by David Woolley)
* ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
with "400 bad request" - DEBUG shows "Received a REFER without a
parseable Refer-To" (Reported by Beppo Mazzucato)
* ASTERISK-24535 - stringfields: Fix regression from fix for
unintentional memory retention and another issue exposed by the
fix (Reported by Corey Farrell)
* ASTERISK-24471 - Crash - assert_fail in libc in
pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
(Reported by yaron nahum)
* ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
in-dialog with invalid target causes crash (Reported by Joshua
Colp)
* ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
module load (Reported by Matt Jordan)
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
allow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24542 - [patch]Failure showing codecs via 'core show
channeltype <tech>' (Reported by snuffy)
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
by xrobau)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24572 - [patch]App_meetme is loaded without its
defaults when the configuration file is missing (Reported by
Nuno Borges)
* ASTERISK-24573 - [patch]Out of sync conversation recording when
divided in multiple recordings (Reported by Nuno Borges)
* ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
reliably transmitted during transfers (Reported by Matt Jordan)
* ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
extension to another pjsip extension (Reported by Abhay Gupta)
Improvements made in this release:
-----------------------------------
* ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
property 'unanswered' (Reported by Matt Jordan)
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
column in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
recordings (Reported by Ben Smithurst)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
lookups (Reported by Birger "WIMPy" Harzenetter)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0-rc1
Thank you for your continued support of Asterisk!
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