On Thu, May 18, 2023 at 12:12 PM Fred Posner <f...@qxork.com> wrote:
> Is this type of message going to be common now? > > Asking as it’s a departure from the lists general traffic. > I sent the test to see what it'd look like and to see what folks thought. The original is written in markdown which looks reasonably good when sent as a text email. The GitHub Action we use to send the email has a "convert markdown to html" option though so I thought I'd try it out. It messed up some of the rendering though so we'll give it a miss. > > > Regards, > > Fred Posner > > > > On May 18, 2023, at 2:07 PM, Asterisk Development Team < > asterisktea...@sangoma.com> wrote: > > > > The Asterisk Development Team would like to announce > > release candidate 1 of Asterisk 20.3.0. > > The release artifacts are available for immediate download at > > https://github.com/asterisk/asterisk/releases/tag/20.3.0-rc1 and > https://downloads.asterisk.org/pub/telephony/asterisk > > This release resolves issues reported by the community > > and would have not been possible without your participation. > > Thank You! > > Change Log for Release 20.3.0-rc1 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 CHERRYPICKTESTINGINPROGRESS > > • .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." > > • respjsipstir_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. > > • bridgebuiltinfeatures: 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 > > • reshttpmedia_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 ASTCELLOCALOPTIMIZEBEGIN can be used by itself or in conert with > the existing ASTCELLOCAL_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. > > • app_senddtmf: Add SendFlash AMI action. > > The SendFlash AMI action now allows sending a hook flash event on a > channel. > > • 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. > > • bridgebuiltinfeatures: add beep via touch variable > > Add optional touch variable : TOUCHMIXMONITORBEEP(interval) Setting > TOUCHMIXMONITORBEEP/TOUCHMONITORBEEP 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. > > • 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. > > • pbx_dundi: Add PJSIP support. > > DUNDi now supports chanpjsip. Outgoing calls using PJSIP require the > pjsipoutgoing_endpoint option to be set in dundi.conf. > > • 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'. > > • reshttpmedia_cache: Introduce options and customize > > The reshttpmediacache module now attempts to load configuration from the > reshttpmediacache.conf file. The following options were added: > > • timeout_secs > > • user_agent > > • follow_location > > • max_redirects > > • protocols > > • redirect_protocols > > • dnscachetimeout_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. > > Upgrade Notes: > > • ### cel: add local optimization begin event The existing > ASTCELLOCALOPTIMIZE can continue to be used as-is and the > ASTCELLOCALOPTIMIZE_BEGIN event can be ignored if desired. > > Closed Issues: > > • #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-rc1.md > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-dev mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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