The Asterisk Development Team has announced the release of Asterisk 12.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.1.0 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:
New Features made in this release:
-----------------------------------
* ASTERISK-23038 - Need config option to enable PJSIP logger at
load time (Reported by Rusty Newton)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23051 - ARI: channel variables in JSON breaks passing
parameters in JSON (Reported by Matt Jordan)
* ASTERISK-22952 - res_pjsip_pubsub: crash when
subscription_destructor is terminated from a non-PJSIP thread
(Reported by Matt Jordan)
* ASTERISK-22486 - ARI: TCP Reset after 204 response (Reported by
David M. Lee)
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
core_event_dispatcher taskprocessor thread (Reported by Etienne
Lessard)
* ASTERISK-23074 - Crash in ChanIsAvail app (Reported by Kilburn)
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
memory when <replace-char> is empty (Reported by Gareth Palmer)
* ASTERISK-22871 - cel_pgsql module not loading after "reload" or
"reload cel_pgsql.so" command (Reported by Matteo)
* ASTERISK-23084 - [patch]rasterisk needlessly prints the
AST-2013-007 warning (Reported by Tzafrir Cohen)
* ASTERISK-23101 - pjsip: crash when parsing scheme from SIP URI
(Reported by Matt Jordan)
* ASTERISK-17138 - [patch] Asterisk not re-registering after it
receives "Forbidden - wrong password on authentication"
(Reported by Rudi)
* ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
lua 5.2 (Reported by George Joseph)
* ASTERISK-23053 - The users of ao2_iterator_cleanup() are
violating the ao2_iterator opacity. (Reported by Richard
Mudgett)
* ASTERISK-22924 - PJSIP MESSAGE support does not present the
contact information on outbound messages (Reported by Anthony
Messina)
* ASTERISK-22884 - hangup_handler end with h extension: tests
currently fail in Asterisk 12 + (Reported by Matt Jordan)
* ASTERISK-23128 - res_ari: Memory leak on response headers and
some JSON response messages (Reported by Joshua Colp)
* ASTERISK-23081 - PJSip Tab Expansion erroring (Reported by
xrobau)
* ASTERISK-22946 - Local From tag regression with sipgate.de
(Reported by Stephan Eisvogel)
* ASTERISK-23065 - On Asterisk start, device state is INVALID for
previously registered PJSIP endpoints, despite re-registrations
(Reported by Rusty Newton)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23034 - [patch] manager Originate doesn't abort on
failed format_cap allocation (Reported by Corey Farrell)
* ASTERISK-23062 - res_pjsip AOR config option qualify_frequency
is inconsistently respected (Reported by Rusty Newton)
* ASTERISK-23071 - pjsip: mailboxes documentation is lacking
(Reported by Matt Jordan)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lain??)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
"transferred" (Reported by Jeremy Lain??)
* ASTERISK-23018 - PJSip 'allow=all' results in failed SDP
negotiation (Reported by xrobau)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23051 - ARI: channel variables in JSON breaks passing
parameters in JSON (Reported by Matt Jordan)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-23177 - [patch] RealTime cant update sipbuddies table
when registering or updating friend (Reported by Denis)
* ASTERISK-23082 - Including g722 in pjsip codec configuration
results in unexpected SDP offers (Reported by xrobau)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-23143 - ARI: subscribing to an already subscribed
resource returns a 500 error (Reported by Matt Jordan)
* ASTERISK-23056 - [patch]INFINITY and NAN undefined (Reported by
capouch)
* ASTERISK-23129 - segfault in res_pjsip_pubsub.so (Reported by
Dan Jenkins)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23106 - pjsip: ACK to 200 OK sent to private IP address
on outbound channel's INVITE request (Reported by Matt Jordan)
* ASTERISK-23072 - MWI subscription from Cisco SPA fails with
PJSIP (Reported by Bob M)
* ASTERISK-23164 - CDRs: mid-call/pre-dial handlers perturb
context/exten/app/data fields during Dial (Reported by Matt
Jordan)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-23249 - Skinny subchannel locking issues (Reported by
Damien Wedhorn)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping "module reload" command (Reported by Gareth Blades)
* ASTERISK-23250 - CDR(start) function is broken due to sizeof
dereference (Reported by snuffy)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23168 - Overriding outbound_auth in a pjsip
registration causes ERROR, assert failure. (Reported by George
Joseph)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
Improvements made in this release:
-----------------------------------
* ASTERISK-22919 - core show channeltypes slicing (Reported by
outtolunc)
* ASTERISK-22868 - chan_pjsip: 'setvar' should be supported on
endpoints (Reported by Joshua Colp)
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
output (Reported by outtolunc)
* ASTERISK-21084 - New SIP Channel Driver - Path Support (Reported
by Matt Jordan)
* ASTERISK-23068 - http: Implement support for chunked
Transfer-Encoding (Reported by Matt Jordan)
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
against libfreeradius-client (Reported by Jeremy Lain??)
* ASTERISK-22984 - ari: Transfer messages not being sent out ARI
WebSocket (Reported by David M. Lee)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0
Thank you for your continued support of Asterisk!
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