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538 lines
24 KiB
HTML
<html><head><title>ChangeLog for asterisk-22.4.0-rc1</title></head><body>
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<h2>Change Log for Release asterisk-22.4.0-rc1</h2>
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<h3>Links:</h3>
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<ul>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0-rc1.html">Full ChangeLog</a> </li>
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<li><a href="https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0-rc1">GitHub Diff</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0-rc1.tar.gz">Tarball</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
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</ul>
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<h3>Summary:</h3>
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<ul>
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<li>Commits: 24</li>
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<li>Commit Authors: 18</li>
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<li>Issues Resolved: 12</li>
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<li>Security Advisories Resolved: 0</li>
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</ul>
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<h3>User Notes:</h3>
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<ul>
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<li>
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<h4>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</h4>
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<p>A Dial timeout on POST /channels/{channelId}/dial will now result in a
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CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
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no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.</p>
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</li>
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<li>
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<h4>contrib: Add systemd service and timer files for malloc trim.</h4>
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<p>Service and timer files for systemd have been added to the
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contrib/systemd/ directory. If you are experiencing memory issues,
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install these files to have "malloc trim" periodically run on the
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system.</p>
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</li>
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<li>
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<h4>Add log-caller-id-name option to log Caller ID Name in queue log</h4>
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<p>This patch adds a global configuration option, log-caller-id-name, to queues.conf
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to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
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When log-caller-id-name=yes, the Caller ID name is included in the queue log,
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Any '|' characters in the caller ID name will be replaced with '_'.
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(provided it’s allowed by the existing log_restricted_caller_id rules).
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When log-caller-id-name=no (the default), the Caller ID name is omitted.</p>
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</li>
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<li>
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<h4>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</h4>
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<p>In cli.conf, you can now define startup commands that run before
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core initialization and before module initialization.</p>
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</li>
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<li>
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<h4>audiosocket: added support for DTMF frames</h4>
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<p>The AudioSocket protocol now forwards DTMF frames with
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payload type 0x03. The payload is a 1-byte ascii representing the DTMF
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digit (0-9,*,#...).</p>
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</li>
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</ul>
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<h3>Upgrade Notes:</h3>
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<ul>
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<li>
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<h4>ARI: REST over Websocket</h4>
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This commit adds the ability to make ARI REST requests over the same
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websocket used to receive events.
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See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</li>
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</ul>
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<h3>Commit Authors:</h3>
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<ul>
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<li>Albrecht Oster: (1)</li>
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<li>Alexei Gradinari: (1)</li>
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<li>Allan Nathanson: (1)</li>
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<li>Andreas Wehrmann: (1)</li>
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<li>Ben Ford: (1)</li>
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<li>Florent CHAUVEAU: (1)</li>
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<li>George Joseph: (4)</li>
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<li>Joshua C. Colp: (1)</li>
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<li>Luz Paz: (1)</li>
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<li>Mark Murawski: (1)</li>
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<li>Mike Bradeen: (1)</li>
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<li>Mkmer: (1)</li>
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<li>Naveen Albert: (3)</li>
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<li>Norm Harrison: (2)</li>
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<li>Peter Jannesen: (1)</li>
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<li>Phoneben: (1)</li>
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<li>Sean Bright: (1)</li>
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<li>Zhai Liangliang: (1)</li>
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</ul>
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<h2>Issue and Commit Detail:</h2>
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<h3>Closed Issues:</h3>
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<ul>
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<li>505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()</li>
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<li>643: [new-feature]: pjsip show contact -- show all details same as AMI PJSIPShowContacts</li>
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<li>963: [bug]: missing hangup cause for ARI ChannelDestroyed when Dial times out</li>
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<li>1091: [improvement]: app queue :add to queue log callerid name</li>
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<li>1144: [bug]: action_redirect don't remove bridge_after_goto data</li>
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<li>1171: [improvement]: Need the capability in audiohook.c for fractional (float) type volume adjustments.</li>
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<li>1181: [bug]: Incorrect PJSIP Endpoint Device States on Multiple Channels</li>
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<li>1190: [bug]: Crash when starting ConfBridge recording over CLI and AMI</li>
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<li>1197: [bug]: ChannelHangupRequest does not show cause code in all cases</li>
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<li>1206: [improvement]: chan_iax2: Minor improvements to documentation and warning messages.</li>
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<li>1220: [bug]: res_pjsip_caller_id: OLI is not parsed if contained in a URI parameter</li>
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<li>1224: [improvement]: app_meetme: Removal version is incorrect</li>
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</ul>
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<h3>Commits By Author:</h3>
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<ul>
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<li>
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<h4>Albrecht Oster (1):</h4>
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</li>
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<li>
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<p>res_pjproject: Fix DTLS client check failing on some platforms</p>
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</li>
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<li>
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<h4>Alexei Gradinari (1):</h4>
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</li>
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<li>
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<p>chan_pjsip: set correct Endpoint Device State on multiple channels</p>
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</li>
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<li>
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<h4>Allan Nathanson (1):</h4>
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</li>
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<li>
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<p>file.c: missing "custom" sound files should not generate warning logs</p>
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</li>
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<li>
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<h4>Andreas Wehrmann (1):</h4>
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</li>
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<li>
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<p>pbx_ael: unregister AELSub application and CLI commands on module load failure</p>
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</li>
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<li>
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<h4>Ben Ford (1):</h4>
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</li>
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<li>
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<p>contrib: Add systemd service and timer files for malloc trim.</p>
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</li>
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<li>
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<h4>Florent CHAUVEAU (1):</h4>
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</li>
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<li>
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<p>audiosocket: added support for DTMF frames</p>
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</li>
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<li>
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<h4>George Joseph (4):</h4>
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</li>
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<li>ARI: REST over Websocket</li>
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<li>ari_websockets: Fix frack if ARI config fails to load.</li>
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<li>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</li>
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<li>
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<p>Prequisites for ARI Outbound Websockets</p>
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</li>
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<li>
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<h4>Joshua C. Colp (1):</h4>
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</li>
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<li>
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<p>channel: Always provide cause code in ChannelHangupRequest.</p>
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</li>
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<li>
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<h4>Luz Paz (1):</h4>
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</li>
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<li>
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<p>docs: Fix typos in apps/</p>
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</li>
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<li>
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<h4>Mark Murawski (1):</h4>
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</li>
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<li>
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<p>chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..</p>
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</li>
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<li>
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<h4>Mike Bradeen (1):</h4>
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</li>
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<li>
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<p>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</p>
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</li>
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<li>
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<h4>Naveen Albert (3):</h4>
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</li>
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<li>chan_iax2: Minor improvements to documentation and warning messages.</li>
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<li>app_meetme: Remove inaccurate removal version from xmldocs.</li>
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<li>
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<p>res_pjsip_caller_id: Also parse URI parameters for ANI2.</p>
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</li>
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<li>
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<h4>Norm Harrison (2):</h4>
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</li>
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<li>audiosocket: fix timeout, fix dialplan app exit, server address in logs</li>
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<li>
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<p>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</p>
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</li>
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<li>
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<h4>Peter Jannesen (1):</h4>
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</li>
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<li>
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<p>action_redirect: remove after_bridge_goto_info</p>
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</li>
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<li>
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<h4>Sean Bright (1):</h4>
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</li>
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<li>
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<p>app_confbridge: Prevent crash when publishing channel-less event.</p>
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</li>
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<li>
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<h4>Zhai Liangliang (1):</h4>
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</li>
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<li>
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<p>Update config.guess and config.sub</p>
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</li>
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<li>
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<h4>mkmer (1):</h4>
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</li>
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<li>
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<p>audiohook.c: Add ability to adjust volume with float</p>
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</li>
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<li>
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<h4>phoneben (1):</h4>
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</li>
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<li>Add log-caller-id-name option to log Caller ID Name in queue log</li>
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</ul>
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<h3>Commit List:</h3>
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<ul>
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<li>res_pjsip_caller_id: Also parse URI parameters for ANI2.</li>
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<li>app_meetme: Remove inaccurate removal version from xmldocs.</li>
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<li>docs: Fix typos in apps/</li>
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<li>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</li>
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<li>chan_iax2: Minor improvements to documentation and warning messages.</li>
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<li>pbx_ael: unregister AELSub application and CLI commands on module load failure</li>
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<li>res_pjproject: Fix DTLS client check failing on some platforms</li>
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<li>Prequisites for ARI Outbound Websockets</li>
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<li>contrib: Add systemd service and timer files for malloc trim.</li>
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<li>action_redirect: remove after_bridge_goto_info</li>
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<li>channel: Always provide cause code in ChannelHangupRequest.</li>
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<li>Add log-caller-id-name option to log Caller ID Name in queue log</li>
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<li>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</li>
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<li>app_confbridge: Prevent crash when publishing channel-less event.</li>
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<li>ari_websockets: Fix frack if ARI config fails to load.</li>
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<li>ARI: REST over Websocket</li>
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<li>audiohook.c: Add ability to adjust volume with float</li>
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<li>audiosocket: added support for DTMF frames</li>
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<li>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</li>
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<li>audiosocket: fix timeout, fix dialplan app exit, server address in logs</li>
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<li>Update config.guess and config.sub</li>
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<li>chan_pjsip: set correct Endpoint Device State on multiple channels</li>
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<li>file.c: missing "custom" sound files should not generate warning logs</li>
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</ul>
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<h3>Commit Details:</h3>
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<h4>res_pjsip_caller_id: Also parse URI parameters for ANI2.</h4>
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<p>Author: Naveen Albert
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Date: 2025-04-26</p>
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<p>If the isup-oli was sent as a URI parameter, rather than a header
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parameter, it was not being parsed. Make sure we parse both if
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needed so the ANI2 is set regardless of which type of parameter
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the isup-oli is sent as.</p>
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<p>Resolves: #1220</p>
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<h4>app_meetme: Remove inaccurate removal version from xmldocs.</h4>
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<p>Author: Naveen Albert
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Date: 2025-04-26</p>
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<p>app_meetme is deprecated but wasn't removed as planned in 21,
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so remove the inaccurate removal version.</p>
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<p>Resolves: #1224</p>
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<h4>docs: Fix typos in apps/</h4>
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<p>Author: Luz Paz
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Date: 2025-04-09</p>
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<p>Found via codespell</p>
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<h4>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</h4>
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<p>Author: Mike Bradeen
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Date: 2025-04-17</p>
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<p>Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs,
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but the Dial command via ARI did not set an explicit reason. This resulted in a
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CANCEL with Normal Call Clearing and corresponding ChannelDestroyed.</p>
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<p>This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the
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other operations.</p>
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<p>Fixes: #963</p>
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<p>UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a
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CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
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no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.</p>
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<h4>chan_iax2: Minor improvements to documentation and warning messages.</h4>
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<p>Author: Naveen Albert
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Date: 2025-04-18</p>
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<ul>
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<li>Update Dial() documentation for IAX2 to include syntax for RSA
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public key names.</li>
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<li>Add additional details to a couple warnings to provide more context
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when an undecodable frame is received.</li>
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</ul>
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<p>Resolves: #1206</p>
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<h4>pbx_ael: unregister AELSub application and CLI commands on module load failure</h4>
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<p>Author: Andreas Wehrmann
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Date: 2025-04-18</p>
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<p>This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out,
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that the AEL module doesn't do proper cleanup when it fails to load.
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This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module()
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returns an error but load_module() doesn't then unregister CLI cmds and the application.</p>
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<h4>res_pjproject: Fix DTLS client check failing on some platforms</h4>
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<p>Author: Albrecht Oster
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Date: 2025-04-10</p>
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<p>Certain platforms (mainly BSD derivatives) have an additional length
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field in <code>sockaddr_in6</code> and <code>sockaddr_in</code>.
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<code>ast_sockaddr_from_pj_sockaddr()</code> does not take this field into account
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when copying over values from the <code>pj_sockaddr</code> into the <code>ast_sockaddr</code>.
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The resulting <code>ast_sockaddr</code> will have an uninitialized value for
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<code>sin6_len</code>/<code>sin_len</code> while the other <code>ast_sockaddr</code> (not converted from
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a <code>pj_sockaddr</code>) to check against in <code>ast_sockaddr_pj_sockaddr_cmp()</code>
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has the correct length value set.</p>
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<p>This has the effect that <code>ast_sockaddr_cmp()</code> will always indicate
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an address mismatch, because it does a bitwise comparison, and all DTLS
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packets are dropped even if addresses and ports match.</p>
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<p><code>ast_sockaddr_from_pj_sockaddr()</code> now checks whether the length fields
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are available on the current platform and sets the values accordingly.</p>
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<p>Resolves: #505</p>
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<h4>Prequisites for ARI Outbound Websockets</h4>
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<p>Author: George Joseph
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Date: 2025-04-16</p>
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<p>stasis:
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* Added stasis_app_is_registered().
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* Added stasis_app_control_mark_failed().
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* Added stasis_app_control_is_failed().
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* Fixed res_stasis_device_state so unsubscribe all works properly.
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* Modified stasis_app_unregister() to unsubscribe from all event sources.
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* Modified stasis_app_exec to return -1 if stasis_app_control_is_failed()
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returns true.</p>
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<p>http:
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* Added ast_http_create_basic_auth_header().</p>
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<p>md5:
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* Added define for MD5_DIGEST_LENGTH.</p>
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<p>tcptls:
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* Added flag to ast_tcptls_session_args to suppress connection log messages
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to give callers more control over logging.</p>
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<p>http_websocket:
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* Add flag to ast_websocket_client_options to suppress connection log messages
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to give callers more control over logging.
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* Added username and password to ast_websocket_client_options to support
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outbound basic authentication.
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* Added ast_websocket_result_to_str().</p>
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<h4>contrib: Add systemd service and timer files for malloc trim.</h4>
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<p>Author: Ben Ford
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Date: 2025-04-16</p>
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<p>Adds two files to the contrib/systemd/ directory that can be installed
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to periodically run "malloc trim" on Asterisk. These files do nothing
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unless they are explicitly moved to the correct location on the system.
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Users who are experiencing Asterisk memory issues can use this service
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to potentially help combat the problem. These files can also be
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configured to change the start time and interval. See systemd.timer(5)
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and systemd.time(7) for more information.</p>
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<p>UserNote: Service and timer files for systemd have been added to the
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contrib/systemd/ directory. If you are experiencing memory issues,
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install these files to have "malloc trim" periodically run on the
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system.</p>
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<h4>action_redirect: remove after_bridge_goto_info</h4>
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<p>Author: Peter Jannesen
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Date: 2025-03-13</p>
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<p>Under certain circumstances the context/extens/prio are stored in the
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after_bridge_goto_info. This info is used when the bridge is broken by
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for hangup of the other party. In the situation that the bridge is
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broken by an AMI Redirect this info is not used but also not removed.
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With the result that when the channel is put back in a bridge and the
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bridge is broken the execution continues at the wrong
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context/extens/prio.</p>
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<p>Resolves: #1144</p>
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<h4>channel: Always provide cause code in ChannelHangupRequest.</h4>
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<p>Author: Joshua C. Colp
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Date: 2025-04-16</p>
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<p>When queueing a channel to be hung up a cause code can be
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specified in one of two ways:</p>
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<ol>
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<li>
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<p>ast_queue_hangup_with_cause
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This function takes in a cause code and queues it as part
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of the hangup request, which ultimately results in it being
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set on the channel.</p>
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</li>
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<li>
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<p>ast_channel_hangupcause_set + ast_queue_hangup
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This combination sets the hangup cause on the channel before
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queueing the hangup instead of as part of that process.</p>
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</li>
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</ol>
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<p>In the #2 case the ChannelHangupRequest event would not contain
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the cause code. For consistency if a cause code has been set
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on the channel it will now be added to the event.</p>
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<p>Resolves: #1197</p>
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<h4>Add log-caller-id-name option to log Caller ID Name in queue log</h4>
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<p>Author: phoneben
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Date: 2025-02-28</p>
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<p>Add log-caller-id-name option to log Caller ID Name in queue log</p>
|
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<p>This patch introduces a new global configuration option, log-caller-id-name,
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to queues.conf to control whether the Caller ID name is logged when a call enters a queue.</p>
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<p>When log-caller-id-name=yes, the Caller ID name is logged
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as parameter 4 in the queue log, provided it’s allowed by the
|
||
existing log_restricted_caller_id rules. If log-caller-id-name=no (the default),
|
||
the Caller ID name is omitted from the logs.</p>
|
||
<p>Fixes: #1091</p>
|
||
<p>UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf
|
||
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
|
||
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
|
||
Any '|' characters in the caller ID name will be replaced with '_'.
|
||
(provided it’s allowed by the existing log_restricted_caller_id rules).
|
||
When log-caller-id-name=no (the default), the Caller ID name is omitted.</p>
|
||
<h4>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</h4>
|
||
<p>Author: George Joseph
|
||
Date: 2025-04-10</p>
|
||
<p>Commands in the "[startup_commands]" section of cli.conf have historically run
|
||
after all core and module initialization has been completed and just before
|
||
"Asterisk Ready" is printed on the console. This meant that if you
|
||
wanted to debug initialization of a specific module, your only option
|
||
was to turn on debug for everything by setting "debug" in asterisk.conf.</p>
|
||
<p>This commit introduces options to allow you to run CLI commands earlier in
|
||
the asterisk startup process.</p>
|
||
<p>A command with a value of "pre-init" will run just after logger initialization
|
||
but before most core, and all module, initialization.</p>
|
||
<p>A command with a value of "pre-module" will run just after all core
|
||
initialization but before all module initialization.</p>
|
||
<p>A command with a value of "fully-booted" (or "yes" for backwards
|
||
compatibility) will run as they always have been...after all
|
||
initialization and just before "Asterisk Ready" is printed on the console.</p>
|
||
<p>This means you could do this...</p>
|
||
<p><code>[startup_commands]
|
||
core set debug 3 res_pjsip.so = pre-module
|
||
core set debug 0 res_pjsip.so = fully-booted</code></p>
|
||
<p>This would turn debugging on for res_pjsip.so to catch any module
|
||
initialization debug messages then turn it off again after the module is
|
||
loaded.</p>
|
||
<p>UserNote: In cli.conf, you can now define startup commands that run before
|
||
core initialization and before module initialization.</p>
|
||
<h4>app_confbridge: Prevent crash when publishing channel-less event.</h4>
|
||
<p>Author: Sean Bright
|
||
Date: 2025-04-07</p>
|
||
<p>Resolves: #1190</p>
|
||
<h4>ari_websockets: Fix frack if ARI config fails to load.</h4>
|
||
<p>Author: George Joseph
|
||
Date: 2025-04-02</p>
|
||
<p>ari_ws_session_registry_dtor() wasn't checking that the container was valid
|
||
before running ao2_callback on it to shutdown registered sessions.</p>
|
||
<h4>ARI: REST over Websocket</h4>
|
||
<p>Author: George Joseph
|
||
Date: 2025-03-12</p>
|
||
<p>This commit adds the ability to make ARI REST requests over the same
|
||
websocket used to receive events.</p>
|
||
<p>For full details on how to use the new capability, visit...</p>
|
||
<p>https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</p>
|
||
<p>Changes:</p>
|
||
<ul>
|
||
<li>Added utilities to http.c:<ul>
|
||
<li>ast_get_http_method_from_string().</li>
|
||
<li>ast_http_parse_post_form().</li>
|
||
</ul>
|
||
</li>
|
||
<li>Added utilities to json.c:<ul>
|
||
<li>ast_json_nvp_array_to_ast_variables().</li>
|
||
<li>ast_variables_to_json_nvp_array().</li>
|
||
</ul>
|
||
</li>
|
||
<li>Added definitions for new events to carry REST responses.</li>
|
||
<li>Created res/ari/ari_websocket_requests.c to house the new request handlers.</li>
|
||
<li>Moved non-event specific code out of res/ari/resource_events.c into
|
||
res/ari/ari_websockets.c</li>
|
||
<li>Refactored res/res_ari.c to move non-http code out of ast_ari_callback()
|
||
(which is http specific) and into ast_ari_invoke() so it can be shared
|
||
between both the http and websocket transports.</li>
|
||
</ul>
|
||
<p>UpgradeNote: This commit adds the ability to make ARI REST requests over the same
|
||
websocket used to receive events.
|
||
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</p>
|
||
<h4>audiohook.c: Add ability to adjust volume with float</h4>
|
||
<p>Author: mkmer
|
||
Date: 2025-03-18</p>
|
||
<p>Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41.</p>
|
||
<p>This is accomplished by the following:
|
||
Convert internal variables to type float.
|
||
Always use ast_frame_adjust_volume_float() for adjustments.
|
||
Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust().
|
||
Cast float to int in ast_audiohook_volume_get()
|
||
Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float.</p>
|
||
<p>This update maintains 100% backward compatibility.</p>
|
||
<p>Resolves: #1171</p>
|
||
<h4>audiosocket: added support for DTMF frames</h4>
|
||
<p>Author: Florent CHAUVEAU
|
||
Date: 2025-02-28</p>
|
||
<p>Updated the AudioSocket protocol to allow sending DTMF frames.
|
||
AST_FRAME_DTMF frames are now forwarded to the server, in addition to
|
||
AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF
|
||
with value 0x03 was added to the protocol. The payload is a 1-byte
|
||
ascii representing the DTMF digit (0-9,*,#...).</p>
|
||
<p>UserNote: The AudioSocket protocol now forwards DTMF frames with
|
||
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
|
||
digit (0-9,*,#...).</p>
|
||
<h4>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</h4>
|
||
<p>Author: Norm Harrison
|
||
Date: 2023-04-03</p>
|
||
<p>Co-authored-by: Florent CHAUVEAU <a href="mailto:florentch@pm.me">florentch@pm.me</a></p>
|
||
<h4>audiosocket: fix timeout, fix dialplan app exit, server address in logs</h4>
|
||
<p>Author: Norm Harrison
|
||
Date: 2023-04-03</p>
|
||
<ul>
|
||
<li>Correct wait timeout logic in the dialplan application.</li>
|
||
<li>Include server address in log messages for better traceability.</li>
|
||
<li>Allow dialplan app to exit gracefully on hangup messages and socket closure.</li>
|
||
<li>Optimize I/O by reducing redundant read()/write() operations.</li>
|
||
</ul>
|
||
<p>Co-authored-by: Florent CHAUVEAU <a href="mailto:florentch@pm.me">florentch@pm.me</a></p>
|
||
<h4>chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..</h4>
|
||
<p>Author: Mark Murawski
|
||
Date: 2025-03-23</p>
|
||
<p>CLI 'pjsip show contact' does not show enough information.
|
||
One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact
|
||
This feature adds the same details as PJSIPShowContacts to the CLI</p>
|
||
<p>Resolves: #643</p>
|
||
<h4>Update config.guess and config.sub</h4>
|
||
<p>Author: Zhai Liangliang
|
||
Date: 2025-03-26</p>
|
||
<h4>chan_pjsip: set correct Endpoint Device State on multiple channels</h4>
|
||
<p>Author: Alexei Gradinari
|
||
Date: 2025-03-25</p>
|
||
<ol>
|
||
<li>
|
||
<p>When one channel is placed on hold, the device state is set to ONHOLD
|
||
without checking other channels states.
|
||
In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN
|
||
to calculate aggregate device state of all active channels.</p>
|
||
</li>
|
||
<li>
|
||
<p>The current implementation incorrectly classifies channels in use.
|
||
The only channels that has the states: UP, RING and BUSY are considered as "in use".
|
||
A channel should be considered "in use" if its state is anything other than
|
||
DOWN or RESERVED.</p>
|
||
</li>
|
||
<li>
|
||
<p>Currently, if the number of channels "in use" is greater than device_state_busy_at,
|
||
the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate
|
||
device state.
|
||
The endpoint device state should be BUSY if the number of channels "in use" is greater
|
||
than or equal to device_state_busy_at.</p>
|
||
</li>
|
||
</ol>
|
||
<p>Fixes: #1181</p>
|
||
<h4>file.c: missing "custom" sound files should not generate warning logs</h4>
|
||
<p>Author: Allan Nathanson
|
||
Date: 2025-03-18</p>
|
||
<p>With <code>sounds_search_custom_dir = yes</code> we first look to see if a sound file
|
||
is present in the "custom" sound directory before looking in the standard
|
||
sound directories. We should not be issuing a WARNING log message if a
|
||
sound cannot be found in the "custom" directory.</p>
|
||
<p>Resolves: https://github.com/asterisk/asterisk/issues/1170</p>
|
||
</body></html>
|