The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.4.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.4.0-rc1 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 candidate:
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-29820 - cli: Add command to evaluate a function
(Reported by N A)
* ASTERISK-29941 - chan_pjsip: Add ability to send flash
events
(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-30007 - chan_iax2: Prevent crashes due to attempted
encryption with missing secrets
(Reported by N A)
* ASTERISK-29990 - chan_dahdi: adding ring cadences is not
idempotent on dahdi restart
(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-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-29994 - chan_dahdi: Round robin array size is too
small for max number of groups
(Reported by N A)
* ASTERISK-29943 - file.c: seeking to negative file offset is
not prevented
(Reported by N A)
* ASTERISK-29843 - Session timers get removed 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-29955 - chan_sip: SIP route header is missing on
UPDATE
(Reported by Mark Petersen)
* ASTERISK-29253 - Incorrect bridging on transfer
(Reported by Yury Kirsanov)
* ASTERISK-29948 - iostream: Infinite TCP timeout writing data
(Reported by N A)
* ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
functionality not enabled
(Reported by Claude Diderich)
* ASTERISK-30006 - res_pjsip: UDP transport does not work when
async_operations is greater than 1
(Reported by Ross Beer)
* ASTERISK-30021 - ast_variable_list_replace_variable uses
variable with new keyword
(Reported by Jasper
Hafkenscheid)
* ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
of SDP attributes
(Reported by Josh Hogan)
* ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
database columns
(Reported by Gregory Massel)
* 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-29950 - SayNumber can handle '01' to '07', but not
'08' or '09'
(Reported by Jim Van Meggelen)
* 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-29961 - RLS: domain part of 'uri' list attribute
mismatch with SUBSCRIBE request
(Reported by Alexei
Gradinari)
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-30008 - samples: Remove obsolete config files
(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-29935 - build: Remove leftover build references
(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-25716 - Documentation: Document explanations and
examples for possible values of DIALSTATUS
(Reported by
Rusty Newton)
* ASTERISK-29980 - build: External binary modules don't use
https
(Reported by INVADE International Ltd.)
* ASTERISK-29967 - pbx_builtins: Add missing documentation
(Reported by N A)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0-rc1
Thank you for your continued support of Asterisk!
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