Never Cross the Streams
Mike Bradeen
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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
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
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
The ability to send SIP NOTIFY messages to endpoints and arbitrary uri’s via AMI and CLI has existed for some time within Asterisk. Changes from
Up until recently Asterisk only supported RFC 4733 RTP events when using 8KHz codecs like G.711. However, with this recent change, Asterisk now supports the
In a previous blog post we talked about using Asterisk’s uni-cast functionality as a bridge between the PSTN and an external service. This post entails
Note: This new implementation is available as of Asterisk 18.22.0, 20.7.0, and 21.2.0. It’s been almost 4 years since STIR/SHAKEN support was first added to
Most SIP endpoints will qualify each other by sending OPTIONS back and forth to make sure they are there and can respond. Asterisk uses this mechanism
With a little over a month before AstriDevCon 2024 this is a reminder that registration is available and required for attendance. You can register by
As you may have seen from a recent submission, it was recently found that there was a consistent delay when adding a channel into the
For the past few yeas, the Asterisk team at Sangoma has been spending way too much time keeping aged Code Review, CI, Issue Tracking and
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