VoiceBlender v0.3.0 is out. 🚀
This version adds support for blind and attended transfers for both originated and incoming calls. These capabilities are standard requirements in call centre environments and voice systems, where efficient call routing and hand-overs must integrate with established SIP and legacy telephony infrastructure. Transfers can be initiated from a SIP device as well as by utilising the VoiceBlender API.
Additional Updates
- Pause recordings — important for handling sensitive data such as credit card numbers.
- Configurable Jitter Buffer — improves quality in poor network conditions.
- Broadcast DTMF to all legs — enables control of external IVRs.
- Join rooms as muted or deaf — avoids race conditions.
- Updated Opus library and added performance tests.
Upgrading
Pull the latest release from GitHub:
go get github.com/VoiceBlender/voiceblender@v0.3.0
go build -o voiceblender ./cmd/voiceblender
Full changelog: v0.2.0…v0.3.0
Check the API Documentation for details on the new transfer and recording endpoints.