After a year of development release of a new stable branch of the open communication platform , used to deploy software PBXs, voice communication systems, VoIP gateways, organizing IVR systems (voice menu), voice mail, telephone conferences and call centers. Project source code licensed under GPLv2.
Asterisk 17 to the category of releases with regular support, for which updates are generated within two years. Support for the previous LTS branch of Asterisk 16 will last until October 2023, and support for the Asterisk 13 branch until October 2021. LTS releases focus on stability and performance optimizations, while regular releases prioritize feature enhancements.
, added in Asterisk 17:
- In ARI (Asterisk REST Interface), an API for creating external communication applications that can directly manipulate channels, bridges and other telephony components in Asterisk, the ability to define event filters is implemented - an application can set a list of allowed or prohibited event types, and after that, applications only events allowed in the white list or not falling under the black list will be transmitted;
- A new 'move' call has been added to the REST API, allowing you to move channels from one application to another without returning to the call handling script (dialplan);
- A new AttendedTransfer application has been added for queuing attended transfers of a call (the operator first connects to the target subscriber himself and after a successful call connects the caller to him) to a given extension number;
- Added a new BlindTransfer application to redirect all channels associated with the caller to the target subscriber ("blind" transfer, when the operator does not know if the callee will answer the call);
- In the ConfBridge conferencing gateway, the "average_all", "highest_all" and "lowest_all" parameters have been added to the remb_behavior option, working at the level of combined channels (bridge), and not at the level of sources, i.e. the REMB (Receiver Estimated Maximum Bitrate) value, which estimates the throughput of the client, is calculated and sent to each sender, and is not tied to a specific sender;
- New variables have been added to the Dial command, designed to establish a new connection and associate it with a channel:
- RINGTIME and RINGTIME_MS - contain the time between the creation of the channel and the receipt of the first RINGING signal;
- PROGRESSTIME and PROGRESSTIME_MS - contain the time between the creation of the channel and the receipt of the PROGRESS signal (equivalent to the value of PDD, Post Dial Delay);
- DIALEDTIME_MS and ANSWEREDTIME_MS are variants of DIALEDTIME and ANSWEREDTIME that return time in milliseconds instead of seconds.
- In rtp.conf for RTP/ICE added the ability to publish the local ice_host_candidate address, as well as the translated address;
- DTLS packets can now be fragmented according to MTU value, allowing larger certificates to be used when negotiating DTLS connections;
- Added "p" option to the ReadExten command to stop reading the extension set after pressing the "#" character;
- Added support for dual binding to IPv4/IPv6 in the DUNDi PBX module;
- For MWI (Message Waiting Indicators), a new module "res_mwi_devstate" has been added, which allows you to subscribe to voice mailboxes using "presence" events, which makes it possible to use the BLF line status keys as voice message waiting indicators;
- The chan_sip driver has been deprecated, instead it is recommended to use the chan_pjsi channel driver built using the SIP stack for the SIP protocol and allows you to get away from the limitations and bottlenecks inherent in the old driver, such as monolithic design, codebase obfuscation, hard-coded restrictions, and laboriousness of adding new features.
Source: opennet.ru