After a year of development release of a new stable branch of the open communication platform , used for deploying software PBXs, voice communication systems, VoIP gateways, organizing IVR systems (voice menus), voicemail, conference calls, and call centers. The source texts of the project is licensed under GPLv2.
Asterisk 17 as regular release categories, with updates being formulated for two years. Support for the previous LTS branch Asterisk 16 will continue until October 2023, while for branch Asterisk 13 until October 2021. When preparing LTS releases, the main focus is on ensuring stability and optimizing performance, whereas the priority for regular releases is on increasing functionality.
, added in Asterisk 17:
- In ARI (Asterisk REST Interface), the API for creating external communication applications that can directly manipulate channels, bridges, and other telephony components in Asterisk, the ability to define event filters has been implemented — the application can specify a list of allowed or denied event types, and after that, only events allowed in the white list or not falling under the black list will be sent to the applications;
- A new call 'move' has been added to the REST API, allowing channels to be moved from one application to another without returning to the call handling script (dialplan);
- A new application AttendedTransfer has been added for queuing attended call transfers (the operator first connects to the target subscriber and after a successful call connects the caller) to a specified extension;
- A new application BlindTransfer has been added for redirecting all channels associated with the caller to the target subscriber ('blind' transfer, where the operator does not know whether the target will answer the call);
- In the conference bridge gateway ConfBridge, new parameters 'average_all', 'highest_all', and 'lowest_all' have been added to the remb_behavior option, which operate on the level of bridged channels, not on the level of sources, i.e., the value of REMB (Receiver Estimated Maximum Bitrate), which estimates the client's bandwidth, is calculated and sent to each sender, and is not tied to a specific sender;
- The Dial command, designed for establishing a new connection and linking it to a channel, has been enhanced with new variables:
- RINGTIME and RINGTIME_MS — contain the time between channel creation and receipt of the first RINGING signal;
- PROGRESSTIME and PROGRESSTIME_MS — contain the time between channel creation and receipt of the PROGRESS signal (equivalent to the PDD, Post Dial Delay);
- DIALEDTIME_MS and ANSWEREDTIME_MS — variants of DIALEDTIME and ANSWEREDTIME that provide time in milliseconds instead of seconds;
- In rtp.conf for RTP/ICE, the ability to publish the local address ice_host_candidate has been added, along with the translated address;
- DTLS packets can now be fragmented according to the MTU value, allowing for larger certificates during DTLS connection negotiations;
- An option 'p' has been added to the ReadExten command to stop reading the extension set after the '#' symbol is pressed;
- The DUNDi PBX module has been enhanced to support dual binding to IPv4/IPv6;
- For MWI (Message Waiting Indicators), a new module 'res_mwi_devstate' has been added, allowing subscriptions to voicemail using 'presence' events, enabling the use of BLF line status keys as voicemail waiting indicators;
- The chan_sip driver has been deprecated, and it is recommended to use the chan_pjsip channel driver for the SIP protocol, which is built using the SIP stack and allows for overcoming the limitations and bottlenecks of the old driver, such as monolithic design, codebase complexity, hardcoded restrictions, and the labor-intensive process of adding new features.
Source: opennet.ru
