VoIP Mailing List Archives

[asteriskteam at digium...]

The Asterisk Development Team has announced the release of Asterisk 12.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.2.0 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-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 Marcello Ceschia)
* 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 Valentinavičius)
* 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, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.2.0

Thank you for your continued support of Asterisk!


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users