833d9b293c
This is a micro update that is mostly security fixes and bug fixes
with very small improvements. In addition to this being a security
fix, asterisk16 is a leaf package.
Upstream changes:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
scenario is causing a crash
(Reported by Gregory Massel)
* ASTERISK-29260 - sRTP Replay Protection ignored; even tears
down long calls
(Reported by Alexander Traud)
* ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
responses causes memory corruption and crash
(Reported by
Ivan Poddubny)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29215 - res_pjsip_session: NULL active_media_state
topology caused asterisk crash
(Reported by sungtae kim)
* ASTERISK-29035 - chan_local: Multistream support breaks T.38
faxing
(Reported by Matthias Hensler)
* ASTERISK-29071 - app_confbridge: Memory rises when
jitterbuffer enabled and muting over AMI occurs
(Reported
by Stefan Ruf)
* ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
if there are multiple progress events
(Reported by N A)
* ASTERISK-24434 - Fix differing usage of assignment operators
in modules.conf
(Reported by Rusty Newton)
* ASTERISK-29306 - strings: Incorrect use of
__attribute__((pure)) in ast_str_to_lower definition
(Reported by Vitezslav Novy)
* ASTERISK-29300 - res_rtp_asterisk: When native local bridging
the remote SSRC becomes permanent
(Reported by Sebastian
Damm)
* ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
REGISTER responses with external_signaling_address
(Reported by Brian Paboojian)
* ASTERISK-29266 - ICE Role conflict with an unauthorized
session
(Reported by Salah Ahmed)
* ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
into progress
(Reported by Sebastian Damm)
* ASTERISK-29297 - say: Y2021 problem – Asterisk cannot say
year 2021 in Dutch
(Reported by Jacek Konieczny)
* ASTERISK-29315 - res_pjsip: re-registration gets stuck if
setting initial auth credentials fails
(Reported by Nick
French)
* ASTERISK-29312 - res_fax: asterisk fails to publish the
Stasis and ReceiveFax status messages if the remote Station ID
contains invalid UTF-8 characters
(Reported by Alexei
Gradinari)
* ASTERISK-16799 - Callee declined when 'beep' audio file does
not exist
(Reported by IAMJames_)
* ASTERISK-29313 - res_pjsip_refer: Segfault in progress
notify
(Reported by George Joseph)
* ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
return one (no more) record
(Reported by Boris P. Korzun)
* ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
(Reported by Benjamin Keith Ford)
* ASTERISK-29311 - res_odbc_transaction sets forcecommit
default value based on isolation level instead of forcecommit
(Reported by Jaco Kroon)
* ASTERISK-28452 - pjsip: <sess-version> of SDP is not
incremented though SDP may be changed on reinvite without SDP
offer
(Reported by Michael Maier)
* ASTERISK-29287 - app.h: C++ compatibility broken
(Reported by Jean Aunis - Prescom)
* ASTERISK-28369 - app_queue: Member device state "invalid"
when second call is ringing and hint is used
(Reported by
Boolah )
* ASTERISK-29203 - res_pjsip_t38: Crash when changing state
(Reported by Gregory Massel)
* ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
making hold/unhold from webrtc client
(Reported by Edvin
Vidmar)
* ASTERISK-29196 - res_pjsip: Segmentation fault
(Reported by Mauri de Souza Meneguzzo (3CPlus))
* ASTERISK-29280 - chan_sip: Allow peers without audio
(text+video).
(Reported by Alexander Traud)
* ASTERISK-29265 - chan_sip: Allow text+video media streams,
again.
(Reported by Alexander Traud)
* ASTERISK-29261 - res_pjsip: user=phone validation fail for
isup numbers containing *#
(Reported by Mark Petersen)
* ASTERISK-29259 - channel: Allow text+video media streams,
again.
(Reported by Alexander Traud)
* ASTERISK-29258 - chan_sip: Audio stream rejected, Other
stream present: Invalid SDP.
(Reported by Alexander Traud)
* ASTERISK-29220 - After T38 reinvite response of 488 a
subsequent G711 reinvite is not processed correctly. Instead the
previous T38 session media is used
(Reported by Robert
Cripps)
* ASTERISK-29248 - res_pjsip_session: res sometimes
uninitialized reported by compiler Clang.
(Reported by
Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-29321 - sorcery: Add support for more intelligent
reloading.
(Reported by Joshua C. Colp)
* ASTERISK-29325 - res_pjsip_registrar: Include source IP
address and port in log messages
(Reported by Joshua C.
Colp)
* ASTERISK-29326 - asterisk: Update copyright/company
(Reported by Joshua C. Colp)
* ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
events
(Reported by Sébastien Duthil)
* ASTERISK-29275 - Support of MIME-type for wav16
(Reported by Boris P. Korzun)
* ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
Transfer (REFER) failure SIP code
(Reported by Dan Cropp)
* ASTERISK-29262 - Support of various URL-schemes by MoH
(Reported by Boris P. Korzun)
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| patches | ||
| DESCR | ||
| distinfo | ||
| Makefile | ||
| options.mk | ||
| PLIST | ||