The new version of Asterisk 17 arrives, the open source VoIP framework

After a year of development the new stable branch of the open communication platform Asterisk 17 was launched, used to implement software PBX, voice communication systems, VoIP gateways, IVR (voice menu) system organization, voice mail, conference calling and call centers.

Asterisk 17 is assigned to the category of releases with regular support, whose updates are being formed within two years. Support for the previous LTS branch of Asterisk 16 will last until October 2023, and the Asterisk 13 branch until October 2021. When preparing the LTS versions, the main objective is to ensure stability and optimize performance, while prioritizing the regular versions is to increase functionality.

Asterisk 17 main news

Among the main novelties that stand out in this new version stands out the inclusion of the new application "BlindTransfer" which provides the function of being able to redirect all the channels associated with the caller to the target in a "blind transfer" mode. This allows the operator to place the call in a mode where it does not know if the person will answer the call.

In the gateway for organizing ConfBridge conferences, the parameters "average_all", "highest_all" and "lower_all" have been added to the option remb_behavior the value REMB (maximum estimated receiver bit rate), which estimates the performance of the client, it is calculated and sent to each sender and is not tied to a specific sender.

ARI (Asterisk REST Interface) the API for creating external communication applications that can directly manipulate channels, bridges and other telephony components in Asterisk, received the improvements to be able to implement the ability to define event filters- The application can specify a list of event types allowed or prohibited, and then in applications only the events allowed in the white list or not in the black list will be transmitted.


As well highlights the addition of a new call to 'move' to the REST API, which allows channels to be transferred from one application to another without going back to the call processing script (dial plan).

A new AttendedTransfer app has been added to queue call transfers accompanied (the operator first connects to the target and after the successful call connects the caller) with the specified extension number.

On the other hand, there is also a new module "res_mwi_devstate" for MWI (Message Waiting Indicators), which allows subscribing to voice mailboxes using the "presence" events, allowing the use of the status keys of the BLF line as voice message waiting indicators.

For "Dial", designed to establish a new connection and your connection to the channel, new variables are added:

  • RINGTIME and RINGTIME_MS: contain the time between the creation of the channel and the reception of the first RINGING signal.
  • PROGRESSTIME and PROGRESSTIME_MS: contain the time between the creation of a channel and the reception of a PROGRESS signal (equivalent to the value of PDD, Post Dial Delay).
  • DIALEDTIME_MS and ANSWEREDTIME_MS: DIALEDTIME and ANSWEREDTIME options, which give time in milliseconds instead of seconds.

Of the other changes that stand out in the ad:

  • In rtp.conf for RTP / ICE, the ability to publish the local ice_host_candidate address as well as the translated address was added.
  • DTLS packets can now be fragmented according to the MTU value, allowing the use of longer certificates when negotiating DTLS connections.
  • Added the "p" option to the ReadExten command to stop reading a set of extensions after pressing the "#" symbol.
  • The DUNDi PBX module adds dual link support to IPv4 / IPv6.

Finally to download the new version of Asterisk 17 they can get it from their website. Or from a terminal by typing the following command:


Or they can install Asterisk from its source code by following the instructions from the following link.

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