[asterisk-users] Asterisk 16.26.0 Now Available

Thu May 12 07:44:09 CDT 2022

The Asterisk Development Team would like to announce the release of Asterisk 16.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.0 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-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-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-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-30015 - pjsip / WebRTC: Chrome creating large number
      of SDP attributes
      (Reported by Josh Hogan)
 * 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-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-29988 - REGRESSION: The build process is requiring
      xmllint or xmlstarlet ro be installed when it shouldn't

      (Reported by George Joseph)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
      wget isn't available
      (Reported by Stefan Ruijsenaars)
 * 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-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-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-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)

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-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-29877 - app_mf: Allow reading a maximum number of
      digits
      (Reported by N A)
 * 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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0

Thank you for your continued support of Asterisk!
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