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* initial WIP to remove freeswitch from media path when not recording or transcribing dial calls * implement release-media and anchor-media operations * mute/unmute now handled by rtpengine * Dial: dtmf detection now based on SIP INFO events from sbcs and rtpengine * add reason to gather action, bugfixes for transcribe and say
2.0 KiB
2.0 KiB