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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
I should just be able to tell it what I want it to do. As the cost comes down for live transcription and the quality
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
If you keep an eye on the Asterisk gitlog, you may have seen some additions to app_voicemail. These changes include the ability to ‘show’ a
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
As part of the transition to Python3 at the end of last year, we introduced a Python Virtual Environment, or venv for the Asterisk TestSuite.
Lately, you may have seen some submissions from Sangoma’s CommUnity team (covered in it’s own post.) Many of these submissions were small but helpful changes,
Asterisk 18.17.0 and 20.2.0 were released recently with support for PJSIP 2.13. This version of PJSIP includes an important change to deal with race conditions
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