{"id":110978,"date":"2023-10-20T09:10:18","date_gmt":"2023-10-20T07:10:18","guid":{"rendered":"https:\/\/prohoster.info\/blog\/novosti-interneta\/reliz-kommunikaczionnoj-platformy-asterisk-21"},"modified":"2023-10-20T09:10:18","modified_gmt":"2023-10-20T07:10:18","slug":"reliz-kommunikaczionnoj-platformy-asterisk-21","status":"publish","type":"post","link":"https:\/\/prohoster.info\/en\/blog\/news\/reliz-kommunikaczionnoj-platformy-asterisk-21","title":{"rendered":"Release of the Asterisk 21 communication platform","gt_translate_keys":[{"key":"rendered","format":"text"}]},"content":{"rendered":"
After a year of development, the release of the new stable branch of the open communication platform Asterisk 21 has occurred, which is used for deploying software PBXs, voice communication systems, VoIP gateways, organizing IVR systems (voice menus), voicemail, teleconferences, and call centers. The source code of the project is available under the GPLv2 license. <\/p>\n
Asterisk 21 is categorized as a standard support release, with updates being provided for two years. Support for the LTS branch Asterisk 20 will last until October 2027, while Asterisk 18 will be supported until October 2025. Support for the LTS branch 17.x has been discontinued. During the preparation of LTS releases, the primary focus is on ensuring stability and optimizing performance, whereas the priority for standard releases is to enhance functionality. <\/p>\n
Changes in Asterisk 21 include: <\/p>\n
\n
The capabilities of the res_pjsip_pubsub module have been expanded, adding additional functionality to the PJSIP SIP stack for distributed data exchange concerning device statuses through the Jabber\/XMPP PubSub extension (sending notifications by subscription).\n
The sig_analog module for analog FXS channels has been enhanced with a capability to hold the called subscriber (CSH, Called Subscriber Held), allowing the user to place an active call on hold, hang up, and then resume the conversation by picking up another phone on the same line. A setting called calledsubscriberheld has been proposed for managing call hold.\n
In the res_pjsip_header_funcs feature, the prefix argument in PJSIP_HEADERS has been made optional (if not specified, all headers will be returned).\n
In the HTTP server (AstHTTP \u2014 AMI over HTTP), the output of the status page has been simplified (the address and port are now displayed in one line).\n
The configuration file users.conf has been declared deprecated.\n
The function ast_gethostbyname() has been declared deprecated, and it should be replaced with the functions ast_sockaddr_resolve() and ast_sockaddr_resolve_first_af().\n
The SLAStation and SLATrunk applications have been migrated from the app_meetme module to app_sla (when using these applications, the modules in modules.conf should be changed).\n
Modules previously declared deprecated have been removed: chan_skinny, app_osplookup, chan_mgcp, chan_alsa, pbx_builtins, chan_sip, app_cdr, app_macro, res_monitor. <\/ul>\n