github asterisk/asterisk 20.3.0
Asterisk Release 20.3.0

latest releases: 21.3.1, 20.8.1, 18.23.1...
13 months ago

Change Log for Release 20.3.0

Summary:

  • Set up new ChangeLogs directory
  • .github: Add AsteriskReleaser
  • chan_pjsip: also return all codecs on empty re-INVITE for late offers
  • cel: add local optimization begin event
  • core: Cleanup gerrit and JIRA references. (#57)
  • .github: Fix CherryPickTest to only run when it should
  • .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
  • .github: Remove separate set labels step from new PR
  • .github: Refactor CP progress and add new PR test progress
  • res_pjsip: mediasec: Add Security-Client headers after 401
  • .github: Add cherry-pick test progress labels
  • LICENSE: Update link to trademark policy.
  • chan_dahdi: Add dialmode option for FXS lines.
  • .github: Update issue templates
  • .github: Remove unnecessary parameter in CherryPickTest
  • Initial GitHub PRs
  • Initial GitHub Issue Templates
  • pbx_dundi: Fix PJSIP endpoint configuration check.
  • Revert "app_queue: periodic announcement configurable start time."
  • res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
  • pbx_dundi: Add PJSIP support.
  • install_prereq: Add Linux Mint support.
  • chan_pjsip: fix music on hold continues after INVITE with replaces
  • voicemail.conf: Fix incorrect comment about #include.
  • app_queue: Fix minor xmldoc duplication and vagueness.
  • test.c: Fix counting of tests and add 2 new tests
  • res_calendar: output busy state as part of show calendar.
  • loader.c: Minor module key check simplification.
  • ael: Regenerate lexers and parsers.
  • bridge_builtin_features: add beep via touch variable
  • res_mixmonitor: MixMonitorMute by MixMonitor ID
  • format_sln: add .slin as supported file extension
  • res_agi: RECORD FILE plays 2 beeps.
  • func_json: Fix JSON parsing issues.
  • app_senddtmf: Add SendFlash AMI action.
  • app_dial: Fix DTMF not relayed to caller on unanswered calls.
  • configure: fix detection of re-entrant resolver functions
  • cli: increase channel column width
  • app_queue: periodic announcement configurable start time.
  • make_version: Strip svn stuff and suppress ref HEAD errors
  • res_http_media_cache: Introduce options and customize
  • main/iostream.c: fix build with libressl
  • contrib: rc.archlinux.asterisk uses invalid redirect.

User Notes:

  • cel: add local optimization begin event

    The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
    by itself or in conert with the existing
    AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

  • chan_dahdi: Add dialmode option for FXS lines.

    A "dialmode" option has been added which allows
    specifying, on a per-channel basis, what methods of
    subscriber dialing (pulse and/or tone) are permitted.
    Additionally, this can be changed on a channel
    at any point during a call using the CHANNEL
    function.

  • pbx_dundi: Add PJSIP support.

    DUNDi now supports chan_pjsip. Outgoing calls using
    PJSIP require the pjsip_outgoing_endpoint option
    to be set in dundi.conf.

  • cli: increase channel column width

    This change increases the display width on 'core show channels'
    amd 'core show channels verbose'
    For 'core show channels', the Channel name field is increased to
    64 characters and the Location name field is increased to 32
    characters.
    For 'core show channels verbose', the Channel name field is
    increased to 80 characters, the Context is increased to 24
    characters and the Extension is increased to 24 characters.

  • app_senddtmf: Add SendFlash AMI action.

    The SendFlash AMI action now allows sending
    a hook flash event on a channel.

  • res_http_media_cache: Introduce options and customize

    The res_http_media_cache module now attempts to load
    configuration from the res_http_media_cache.conf file.
    The following options were added:

    • timeout_secs
    • user_agent
    • follow_location
    • max_redirects
    • protocols
    • redirect_protocols
    • dns_cache_timeout_secs
  • test.c: Fix counting of tests and add 2 new tests

    The "tests" attribute of the "testsuite" element in the
    output XML now reflects only the tests actually requested
    to be executed instead of all the tests registered.
    The "failures" attribute was added to the "testsuite"
    element.
    Also added two new unit tests that just pass and fail
    to be used for testing CI itself.

  • res_mixmonitor: MixMonitorMute by MixMonitor ID

    It is now possible to specify the MixMonitorID when calling
    the manager action: MixMonitorMute. This will allow an
    individual MixMonitor instance to be muted via ID.
    The MixMonitorID can be stored as a channel variable using
    the 'i' MixMonitor option and is returned upon creation if
    this option is used.
    As part of this change, if no MixMonitorID is specified in
    the manager action MixMonitorMute, Asterisk will set the mute
    flag on all MixMonitor audiohooks on the channel. Previous
    behavior would set the flag on the first MixMonitor audiohook
    found.

  • bridge_builtin_features: add beep via touch variable

    Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
    Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
    interval in seconds will result in a periodic beep being
    played to the monitored channel upon MixMontior/Monitor
    feature start.
    If an interval less than 5 seconds is specified, the interval
    will default to 5 seconds. If the value is set to an invalid
    interval, the default of 15 seconds will be used.

  • format_sln: add .slin as supported file extension

    format_sln now recognizes '.slin' as a valid
    file extension in addition to the existing
    '.sln' and '.raw'.

Upgrade Notes:

  • cel: add local optimization begin event

    The existing AST_CEL_LOCAL_OPTIMIZE can continue
    to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
    can be ignored if desired.

Closed Issues:

  • #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
  • #39: [Bug]: Remove .gitreview from repository.
  • #43: [Bug]: Link to trademark policy is no longer correct
  • #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
  • #52: [improvement]: Add local optimization begin cel event

For more details, see:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md

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