After a year of development, the new stable branch of the open communication platform Asterisk 19 has been released, used for deploying software PBXs, voice communication systems, VoIP gateways, organizing IVR systems (voice menus), voicemail, phone conferences, and call centers. The source texts of the project are available under the GPLv2 license.
Asterisk 19 is classified as a regular support release, with updates provided for two years. Support for the previous LTS branch Asterisk 18 will last until October 2025, and for the Asterisk 16 branch until October 2023. Support for the LTS branch 13.x and the intermediate branch 17.x has been discontinued. When preparing LTS releases, the main focus is on ensuring stability and optimizing performance, while the priority for regular releases is to enhance functionality.
Key improvements in Asterisk 19:
- Debug log categories have been implemented, allowing configuration of the output to show only the necessary debug information. Currently, the following categories are offered: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, and stun_packet.
- A new log formatting mode 'plain' has been added, where logs are output without unnecessary control characters (without highlighting), showing the name of the file, function, and line number. It also provides the ability to define custom logging levels and change the date and time output format in logs.
- The AMI (Asterisk Manager Interface) has been enhanced with the capability to attach handlers for events related to the receipt of tonal signals (DTMF) 'flash' (brief channel interruption).
- In the Originate command (call initiation), a feature has been implemented to set variables for the new channel.
- In the SendMF command and PlayMF manager, support for sending arbitrary R1 MF (multi-frequency) tonal signals to any channel has been added.
- In the MessageSend command, the option to separately specify the 'Destination' and 'To' addresses has been provided.
- A ConfKick command has been added, allowing the disconnection of a specific channel from a conference, all users, or users without administrator rights.
- A Reload command has been introduced for reloading modules.
- A WaitForCondition command has been added to pause the execution of the call processing script (dialplan) until certain conditions are met.
- The app_dial module has added an option 'A' that allows playing sound during a call for both the calling and the called party.
- The app_dtmfstore module has been added, which stores the entered digits of the DTMF tone in a variable.
- The app_morsecode module has implemented support for the American dialect of Morse code and provides a setting to change the pause interval.
- In the app_originate module, for calls initiated from dialplan scripts, the option to specify codecs, call files, and control actions has been added.
- In the app_voicemail module, the ability to send greetings and voicemail usage instructions early and to create a channel only after the time for recording an incoming message has arrived has been added.
- The astcachedir setting has been added to change the location of the cache on disk. By default, the cache is now located in a separate directory /var/cache/asterisk instead of the /tmp directory.
After three years of development, the release of FreePBX 16 has been published, which enhances the web interface for managing Asterisk and provides a ready-made distribution for quickly deploying VoIP systems. Notable changes include support for PHP 7.4, an expanded API based on the GraphQL query language, a transition to a unified PJSIP driver (with the Chan_SIP driver disabled by default), and support for creating templates to modify user interface design. control panel, a redesigned firewall module with enhanced capabilities for managing SIP traffic, the ability to configure protocol parameters for HTTPS, binding AMI to localhost by default, and an option to check password strength.
Also noteworthy is a corrective update to the FreeSWITCH 1.10.7 VoIP telephony platform, which fixes 5 vulnerabilities that could lead to the sending of SIP messages without authentication (e.g., for spoofing and spam distribution through a SIP gateway), leaking session authentication hashes, and carrying out DoS attacks (exhausting free memory and crashes) to block operation. server by sending invalid SRTP packets or flooding with SIP packets.
Source: opennet.ru
