After a year of development the release of a stable branch of the open communication platform Asterisk 24 has taken place, used for the deployment of software PBXs, voice communication systems, VoIP gateways, organization of IVR systems (voice menu), voice mail, telephone conferences and call centers. The project’s source code is written in C and is available under the GPLv2 license.
Asterisk 24 is classified as a Long Term Support (LTS) release, for which updates will be released over a period of five years instead of the usual two years for releases. Support for the previous LTS branch of Asterisk 22 will last until October 2029, and support for the Asterisk 20 branch until October 2027. LTS releases focus on stability and performance optimization, while regular releases focus on increasing functionality.
- In module res_pjsip, which provides support for SIP-based VoIP communications, PJSIP contacts associated with DNS names are now considered accessible if at least one associated DNS SRV record has been resolved, A or AAAA.
- Options “channels_detail_metrics_enabled” and “bridges_detail_metrics_enabled” have been added to the res_prometheus module to control the export of additional metrics channels (‘asterisk_channels_state’, ‘asterisk_channels_duration_seconds’) and bridges (‘asterisk_bridges_channels_count’).
- The “announcements” option has been added to the bridge profile for organizing ConfBridge conferences, with which you can disable general conference announcements, which saves one channel and one taskprocessor thread.
- The chan_websocket module implements an unbuffered mode that sends frames as they are received from the WebSocket and the application.
- Added configuration examples for ITSP IP authentication and use of TLS with SRTP.
- Added “write_timeout” parameter to websocket_client.conf to limit the maximum time a write operation can take before returning an error.
- Added script to res_geolocation tools that uses Alembic to create tables geoloc_location and geoloc_profile.
- The write_timeout field has been added to the ast_websocket_client_options structure to configure the timeout for write operations in non-blocking mode.
- The passthrough and unbuffered fields have been added to the MEDIA_START event in chan_websocket, reflecting the channel creation mode.