Asterisk is a free software program that provides features of a telephone exchange
Announced the launch of the new version of Asterisk 21, which is promoted as a regular release, with updates implemented over a period of two years, which differs from LTS versions, such as the previous Asterisk 20 version, these LTS versions have 5-year support.
For those who do not know about Asterisk, you should know that This is used to implement software PBX, voice communication systems, VoIP gateways, organize IVR (voice menu) systems, voice mail, conference calls and call centers.
Asterisk 21 main news
In this new version of Asterisk 21, various improvements have been integrated into the modules, their capabilities have been expanded, as well as modules and functions have been eliminated.
And in Asterisk 21, on the part of the module improvements highlights the improvements in the module res_pjsip_pubsub, in which its capabilities have been expanded when adding additional SIP stack to PJSIP for distributed data exchange device status via the Jabber/XMPP PubSub extension (sending notifications via subscription), as well as adding body_type to test_handler for unit testing
It is mentioned that the existing res_pjsip_pubsub APIs are somewhat limited in what they can do, but thanks to the additional capabilities implemented, they have gained more feature-rich functions.
Another module that received improvements in this new version is the module sig_analog for analog FXS channels, which iinclude the CSH feature, which allows the user to put an initiated call on hold, hang up and resume the conversation by lifting the handset of another telephone on the same line. To manage call retention, a configuration called subscriber on hold is proposed.
In addition to this, now the applications SLAStation and SLATrunk have been moved from the app_meetme module to app_sla, as well as that the users.conf configuration is now deprecated and will be removed in a future version of Asterisk.
Regarding the modules that have become obsolete in this new version, mention is made of res_monitor, which has been deprecated since Asterisk 16 and is now being removed in accordance with
the Asterisk module deprecation policy. This also removes the 'w' and 'W' options for app_queue. With this, MixMonitor should now be the default and only option for all configurations that previously used Monitor or MixMonitor.
app_macro is another of the modules that were removed, since it was declared deprecated in Asterisk 16. It is mentioned that for most modules that interacted with app_macro,
This change is limited to stopping searching the current context of the macrocontext when it is set and mentioning that The change has additional impacts on the following modules: app_dial which is no longer supported by the connected/redirected M^ macro, app_minivm in which samples written with macro will no longer work, app_queue in which a macro can no longer be called (it is recommended to use gosub which is currently supported),
Of the other changes that stand out of this new version:
- The ast_gethostbyname() function has been deprecated and should be replaced by the ast_sockaddr_resolve() and ast_sockaddr_resolve_first_af() functions.
- In the res_pjsip_header_funcs function, the prefix argument in PJSIP_HEADERS has been made optional (if not specified, all headers will be returned).
- On the http server (AstHTTP – AMI over HTTP), the display of the status page has been simplified (address and port are now displayed on one line).
- The app_osploo module was removed, as it has been declared obsolete in Asterisk 19
- The chan_alsa module was removed, as it has been declared obsolete in Asterisk 19..
- The chan_sip module was removed, as it has been declared obsolete in Asterisk 17.
- app_audiosocket: Fixed timeout with -1 to avoid a busy loop.
- download_externals: fix some version related issues
Finally if you are interested in knowing more about it, you can check the details in the following link