[asterisk-dev] Asterisk 12.2.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Fri Mar 28 15:14:09 CDT 2014
The Asterisk Development Team has announced the first release candidate of
Asterisk 12.2.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.2.0-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-23276 - Look at adding the 'pjsip show channel' command
(Reported by George Joseph)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23290 - chan_sip: ast_bridge_transfer_blind causes
channel to be hung up immediately, leading to BYE request being
sent before NOTIFY (Reported by Matt Jordan)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by marcelloceschia)
* ASTERISK-23125 - ARI: URI is case sensitive (Reported by Zane
Conkle)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-22738 - "Security denial" error in calls from H323
trunk (ooh323.c) (Reported by Gabriele Odone)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-23266 - [patch]pjsip_cli: Memory leak in
ast_sip_cli_print_sorcery_objectset (Reported by George Joseph)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning "Don't know how to indicate
condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23320 - Preloading pbx_config.so with a CustomPresence
hint defined results in crash (Reported by xrobau)
* ASTERISK-23265 - Preloading Certain Modules in Asterisk 12
causes a core dump (Reported by Andrew Nagy)
* ASTERISK-23287 - res_pjsip_refer: Crash during attended transfer
when attended->transferer_second channel is NULL (Reported by
Matt Jordan)
* 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)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
ibercom)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis ValentinaviÄius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
(Reported by John)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
to yes (Reported by Alexandr Gordeev)
* ASTERISK-23258 - Target_uri for LiveRecording model in ARI
(Reported by Ben Merrills)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-23204 - Device state cache requires improvement
(Reported by Mark Michelson)
* ASTERISK-23092 - cli: pjsip show endpoint <endpoint> shows
allow/disallow codecs the same (Reported by Dan Jenkins)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23210 - Security: Remote crash in res_pjsip. (Reported
by Joshua Colp)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23020 - PJSip - Multihomed machine returning wrong IP
address (Reported by xrobau)
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
leaving Conference (Reported by Benjamin Keith Ford)
* ASTERISK-23295 - ARI: ChannelEnteredBridge event not delivered
to client during bridge move operation (Reported by Matt Jordan)
* ASTERISK-23444 - Playback and Record events not subscribed to
when calling Play/Record on bridge (Reported by Ben Merrills)
* ASTERISK-23235 - pjsip transport/tos interpreted differently
than endpoint/tos_audio (Reported by George Joseph)
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter
function in manager.c (Reported by Etienne Lessard)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* ASTERISK-23461 - Only first user is muted when joining
confbridge with 'startmuted=yes' (Reported by Chico Manobela)
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
request (Reported by Kelly Goedert)
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
at astobj2.c:120 (Reported by Jamuel Starkey)
* ASTERISK-23254 - Bad ao2_find() usage in pjsip_options.c
(Reported by Richard Mudgett)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
play empty files for numbers divisible by 100 (Reported by
zvision)
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
(Reported by JoshE)
* ASTERISK-23391 - Audit dialplan function usage of channel
variable (Reported by Corey Farrell)
* ASTERISK-23548 - POST to ARI sometimes returns no body on
success (Reported by Scott Griepentrog)
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly
and gatekeeper is not available (Reported by Dmitry Melekhov)
Improvements made in this release:
-----------------------------------
* ASTERISK-22537 - Create Sorcery equivalent to the AST_CONFIG
function (Reported by George Joseph)
* ASTERISK-23275 - CLI command 'pjsip show registrations' missing
(Reported by George Joseph)
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
not have a call in progress (Reported by Chris Hillman)
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
function to read the whole available data at first and then wait
for any fragmented packets (Reported by Thava Iyer)
* ASTERISK-23233 - alembic missing scripts for certain realtime
tables (Reported by jmls)
* ASTERISK-22537 - Create Sorcery equivalent to the AST_CONFIG
function (Reported by George Joseph)
* ASTERISK-23120 - ARI/AMI: allow objects created via interfaces
to have their unique ID specified by the external application
(Reported by Matt Jordan)
* ASTERISK-22008 - Config framework docs should display the
see-also information in CLI output. (Reported by Richard
Mudgett)
* ASTERISK-23435 - PJSIP: Fix the DNS resolution (whoops)
(Reported by Matt Jordan)
* ASTERISK-22499 - ARI documentation - point to HTTP server
configuration sample and wiki docs where appropriate (Reported
by Rusty Newton)
* ASTERISK-23437 - ARI: Add the ability to add properties to a
bridge on creation (Reported by Matt Jordan)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.2.0-rc1
Thank you for your continued support of Asterisk!
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