The Asterisk Development Team would like to announce the release of Asterisk
18.12.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities
(Reported by Clint Ruoho)
* ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
terminating \
(Reported by Leandro Dardini)
* ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
large files
(Reported by Benjamin Keith Ford)
New Features made in this release:
-----------------------------------
* ASTERISK-29931 - Option to allow a user to not hear the join
sound on enter but everyone else can
(Reported by Michael
Cargile)
* ASTERISK-29968 - func_db: Add a function to return
cardinality of keys at prefix
(Reported by N A)
* ASTERISK-29486 - Hint-like extension value lookup function
without device state
(Reported by N A)
* ASTERISK-29941 - chan_pjsip: Add ability to send flash
events
(Reported by N A)
* ASTERISK-29820 - cli: Add command to evaluate a function
(Reported by N A)
* ASTERISK-29876 - app_queue: Add music on hold option
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29655 - res_pjsip_session: No video to caller if no
camera available
(Reported by Michael Auracher)
* ASTERISK-29638 - res_pjsip_session: No video after early
media
(Reported by Michael Auracher)
* ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
when Picking Up Dahdi Call On Hold
(Reported by Josh
Alberts)
* ASTERISK-29990 - chan_dahdi: adding ring cadences is not
idempotent on dahdi restart
(Reported by N A)
* ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
encryption with missing secrets
(Reported by N A)
* ASTERISK-29728 - menuselect: Disabled by default modules that
are enabled are always recompiled
(Reported by N A)
* ASTERISK-30002 - app_meetme: Don't erroneously set global
variables when channel is NULL
(Reported by N A)
* ASTERISK-29994 - chan_dahdi: Round robin array size is too
small for max number of groups
(Reported by N A)
* ASTERISK-22246 - Asterisk's "T" flag is ignored when used
with "r" or "R" flags. (documentation bug)
(Reported by
Rusty Newton)
* ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
for "disable console colorization"
(Reported by Sebastian
Gutierrez)
* ASTERISK-29843 - Session timers get removed on UPDATE
(Reported by Mark Petersen)
* ASTERISK-29943 - file.c: seeking to negative file offset is
not prevented
(Reported by N A)
* ASTERISK-29955 - chan_sip: SIP route header is missing on
UPDATE
(Reported by Mark Petersen)
* ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
even if early_media already enabled
(Reported by Mark
Petersen)
* ASTERISK-29948 - iostream: Infinite TCP timeout writing data
(Reported by N A)
* ASTERISK-29253 - Incorrect bridging on transfer
(Reported by Yury Kirsanov)
* ASTERISK-30006 - res_pjsip: UDP transport does not work when
async_operations is greater than 1
(Reported by Ross Beer)
* ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
functionality not enabled
(Reported by Claude Diderich)
* ASTERISK-30021 - ast_variable_list_replace_variable uses
variable with new keyword
(Reported by Jasper
Hafkenscheid)
* ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
database columns
(Reported by Gregory Massel)
* ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
of SDP attributes
(Reported by Josh Hogan)
* ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
Disconnecting channel for lack of RTP activity
(Reported
by Dmitriy Serov)
* ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
lack of RTP activity in one way sessions
(Reported by
Boris P. Korzun)
* ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
(Reported by LA)
* ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
(Reported by Daniel Bonazzi)
* ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
context (AST_PBX_MAX_STACK - 1)
(Reported by Tzafrir
Cohen)
* ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
wget isn't available
(Reported by Stefan Ruijsenaars)
* ASTERISK-29988 - REGRESSION: The build process is requiring
xmllint or xmlstarlet ro be installed when it shouldn't
(Reported by George Joseph)
* ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
show netstats printout
(Reported by N A)
* ASTERISK-29939 - agi: Fix xmldoc bug with set music
(Reported by N A)
* ASTERISK-28891 - documentation: AGICommand_set+music
documentation arguments displayed incorreclty
(Reported by
Jonathan Harris)
* ASTERISK-29048 - chan_iax2: "iax2 show registry" shows host
for perceived
(Reported by David Herselman)
* ASTERISK-29674 - Adjust for 64bit time_t
(Reported by
Andre Heider)
* ASTERISK-29961 - RLS: domain part of 'uri' list attribute
mismatch with SUBSCRIBE request
(Reported by Alexei
Gradinari)
* ASTERISK-29928 - logging messages truncated when using MUSL
runtime
(Reported by Philip Prindeville)
* ASTERISK-29960 - ari: Retrieving stored recording can returns
wrong file
(Reported by Arix)
* ASTERISK-29950 - SayNumber can handle '01' to '07', but not
'08' or '09'
(Reported by Jim Van Meggelen)
Improvements made in this release:
-----------------------------------
* ASTERISK-24827 - Missing documentation for chan_dahdi dial
string ring cadences
(Reported by Scott Griepentrog)
* ASTERISK-29940 - general: Add since tags to xmldocs
(Reported by N A)
* ASTERISK-29726 - Add Asterisk External Application Protocol
(AEAP) implementation
(Reported by Kevin Harwell)
* ASTERISK-29951 - app_mf, app_sf: Return -1 on hangup
(Reported by N A)
* ASTERISK-29954 - app_meetme: Emit warning if conference not
found
(Reported by N A)
* ASTERISK-29351 - Qualify pjproject 2.12 for Asterisk
(Reported by George Joseph)
* ASTERISK-29976 - Should Readme include information about
install_prereq script?
(Reported by Marcel Wagner)
* ASTERISK-29970 - Use pkg-config to find libxml2 headers and
libraries
(Reported by Hugh McMaster)
* ASTERISK-29980 - build: External binary modules don't use
https
(Reported by INVADE International Ltd.)
* ASTERISK-25716 - Documentation: Document explanations and
examples for possible values of DIALSTATUS
(Reported by
Rusty Newton)
* ASTERISK-29967 - pbx_builtins: Add missing documentation
(Reported by N A)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
Thank you for your continued support of Asterisk!
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