Approaches to Transcription
Mike Bradeen
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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
Blog Author
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
As we merge in a number of positive CommUnity UCaaS submissions, it is a good reminder of the challenges associated with private forks and forks
Asterisk 21 is scheduled to become the next major release. As part of that process there are a number of modules being removed, see: wiki.asterisk.org.
chan_sip will no longer be included with Asterisk as of the release of version 21. Deprecated in version 17, chan_sip has been scheduled for removal for
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