After a year of development, a new stable branch of the open communication platform Asterisk 19 was released, used to deploy software PBXs, voice communication systems, VoIP gateways, organizing IVR systems (voice menu), voice mail, conference calls and call centers. The source texts of the project are available under the GPLv2 license.
Asterisk 19 is categorized as a regular support release, with updates being rolled out over a two-year period. Support for the previous Asterisk 18 LTS branch will continue until October 2025, and for the Asterisk 16 branch until October 2023. The 13.x LTS branch and the 17.x intermediate branch have been discontinued. LTS releases focus on stability and performance optimizations, while regular releases prioritize feature enhancements.
Key improvements in Asterisk 19:
- Implemented categories of debug logs that allow you to customize the output of only the necessary debug information. The currently suggested categories are dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, and stun_packet.
- A new log formatting mode "plain" has been added, in which the file name, function and line with a number are displayed in the log without unnecessary control characters (without highlighting). It is also possible to define your own levels of logging and change the format of output to the log of dates and times.
- The AMI (Asterisk Manager Interface) adds the ability to attach handlers for events associated with the arrival of a tonal signal (DTMF) "flash" (short-term channel break).
- In the Originate command (call initiation), the ability to set variables for a new channel is implemented.
- Support for sending arbitrary R1 MF (multi-frequency) tones to any channel has been added to the SendMF command and the PlayMF manager.
- The MessageSend command provides the ability to separately specify destination addresses "Destination" and "To".
- Added the ConfKick command, which allows you to disconnect a specific channel, all users or users without administrator rights from the conference.
- Added Reload command to reload modules.
- A WaitForCondition command has been added to suspend the execution of a call processing script (dialplan) until certain conditions are met.
- Added "A" option to the app_dial module to play sound for both the caller and the callee during a call.
- The app_dtmfstore module has been added, which saves dialed tone dialing digits in a variable.
- The app_morsecode module implements support for American Morse code and provides a setting to change the pause interval.
- In the app_originate module for calls initiated from dialplan scripts, the ability to specify codecs, call files, and control actions has been added.
- In the app_voicemail module, the ability to send a greeting and instructions for using voicemail early and create a channel only after it is time to record an incoming message has been added.
- Added setting astcachedir to change cache location on disk. By default, the cache is now located in a separate /var/cache/asterisk directory instead of the /tmp directory.
At the same time, after three years of development, the release of the FreePBX 16 project was published, which develops a web interface for managing Asterisk and a ready-made distribution kit for quickly deploying VoIP systems. Changes include support for PHP 7.4, API extension based on the GraphQL query language, transition to a single PJSIP driver (Chan_SIP driver is disabled by default), support for creating templates to change the design of the user control panel, a redesigned firewall module with advanced features for managing SIP- traffic, the ability to configure protocol parameters for HTTPS, binding AMI only to localhost by default, an option to check the strength of passwords.
You can also note the corrective update of the FreeSWITCH 1.10.7 VoIP telephony platform, which fixes 5 vulnerabilities that can lead to sending SIP messages without authentication (for example, for spoofing and spamming through the SIP gateway), leakage of session authentication hashes and the implementation of DoS attacks (out of free memory and crashes) to block server operation by sending incorrect SRTP packets or flooding SIP packets.
Source: opennet.ru