Exchanging media between Asterisk and external apps has fairly involved for a few reasons, the main one being that RTP, the primary protocol for exchanging
An AstriDevCon video conference was held on July 20, 2025 with about 15 attendees. You can get a recap of the event at AstriDevCon July 2025
If you were able to join us for Astricon in February 2025, you may have caught a Voice AI demo using external media given by
In a previous post we saw an example of how to use Asterisk’s audiohooks to build an external media bridge using ARI. In this post
Historically, using ARI required using HTTP for making REST requests and getting their responses, and a Websocket for receiving events. With Asterisk 20.14.0, 21.9.0 and
If you are new to audio over websocket and come from the telephony world, you might assume that this works similarly to an RTP stream
One of the historical issues we’ve had with Asterisk and the Asterisk Testsuite has been that if you’ve created an Asterisk pull request for some
If you didn’t already know, both SIP and HTTP share the same digest authentication mechanism described all the way back in RFC-2069 “An Extension to HTTP
Overview When recording audio, it can be useful to split streams from one another. For example, if you are in a call with someone, you
As mentioned in this post, Asterisk now supports the use of RFC4733 digits in common bitrates beyond 8kHz. At the end of the post, we
