After a year of development, a new stable branch of the open communication platform Asterisk 18, which used to implement software PBX, voice communication systems, VoIP gateways, host IVR systems (voice menu), voicemail, conference calls and call centers and that its project source code is available under the GPLv2 license.
Perhaps the most interesting thing about Asterisk is that recognizes many VoIP protocols such as SIP, H.323, IAX and MGCP. Asterisk can interoperate with IP terminals acting as a registrar and as a gateway between the two. One of the strengths of the Asterisk software is that it allows the unification of technologies: VoIP, GSM and PSTN.
Asterisk 18 main news
In this new version added support for STIR / SHAKEN protocols to combat caller ID spoofing. With these new added protocols it is supported both the sending of a header with identity guarantee for outgoing calls and the verification of the caller when receiving incoming calls with identity certification for outgoing calls, and verification of the caller when receiving incoming calls.
Another change that is presented in this new version of Asterisk 18, is that added a new 'simple' record formatting mode, It does not use control characters for highlighting and provides information about the file, the function, and the line number.
Options were added "Maximum_sample_rate" to the ConfBridge conferencing gateway to set the maximum sample rate and options "Text_messaging" to control whether the user has text messaging enabled.
In ARI (Asterisk REST Interface), an API for creating external communication applications that can directly manipulate channels, bridges and other telephony components in Asterisk, added parameter 'inhibitConnectedLineUpdates' for calls to 'bridges.addChannel' to prevent the ID of a new connected line from being passed to others combined channel participants. The »externalMedia» subresource has been added to the ARI Channel resource, with the help of which you can replace the sound of an external server on the combined channels or transmit the sound of the combined channels to an external server.
The behavior of the BridgeAdd application is similar to that of the Bridge application and it also sets the BRIDGERESULT variable for the channel, so that the information about the result of the channel combination is passed to the call processing script (dialplan).
The res_pjsip module implements new options Input_call_offer_pref and outgoing_call_offer_pref to define the desired order of codecs for incoming and outgoing calls.
AMI (Asterisk Manager Interface) added the ability to specify 'Content-Type' for SendText actions.
Of the other changes that stand out in this new version:
- For applications and channels, support for the AudioSocket bidirectional audio transmission protocol was implemented.
- The configuration "hide_messaging_ami_events»Is enabled by default to exclude messaging events to reduce the load on AMI and ARI applications.
- Added support for the H.265 / HEVC video codec.
- The Dial, Page, and ChanIsAvail applications allow the use of empty positions in the mailing list, simplifying call processing scenarios by eliminating the need to search for empty positions.
- Added option «enable_status»To the built-in http server to disable the processing of the internal page« / httpstatus ».
- Added "playlist" mode to res_musiconhold, allowing you to specify a list of files or URLs for playback.
- Res_rtp_asterisk has converted the blacklist engine to an access list system (ACL) with the options ice_deny, ice_permit, ice_acl, stun_deny, stun_permit, and stun_acl.
- The Streams API implements the basic capabilities for managing codec negotiation (ACN, advanced codec negotiation).
Finally if you want to know more about it about this new version, you can check the details In the following link.
As for the packages of this new version, you can find them In the following link.