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

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
 * 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
      (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
      (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
      when Picking Up Dahdi Call On Hold
      (Reported by Josh
 * 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
 * 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
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
      (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
 * 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
      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
      (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
      context (AST_PBX_MAX_STACK - 1)
      (Reported by Tzafrir
 * 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
      (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

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
      (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
      (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
      (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:

Thank you for your continued support of Asterisk!
-- Bandwidth and Colocation Provided by --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:

Reply via email to