Vodia PBX now offers real-time call transcription through a seamless integration with Whisper AI, OpenAI’s advanced speech recognition system. With support for multiple languages, technical vocabulary, and noisy environments, Whisper delivers accurate transcriptions even in complex call scenarios. Administrators can enable transcription per tenant using an OpenAI API key, making setup simple and flexible. Once active, all calls are automatically transcribed and accessible in the user portal for easy review and record-keeping. This powerful integration brings enhanced clarity, compliance, and insight to voice communication—whether you're managing support teams, analyzing conversations, or working across language barriers.
Vodia is pleased to announce another enhancement to our phone system - call transcription with Whisper AI. This seamless integration enables Vodia customers to deploy Whisper AI speech-to-text capability within their communications ecosystem, and to configure it for individual tenants.
Whisper, developed by OpenAI, is an Automatic Speech Recognition (ASR) system. Trained on nearly three quarters of a million hours of supervised multilingual and multitask data from the Web, Whisper deftly handles technical language, background noises, and accents (thanks to this diverse, massive data set). It transcribes in a multitude of languages and translates all of them into English.
The Whisper architecture, an encoder-decoder transformer, is a simple, stem-to-stern approach: it separates audio into 30-second segments, which are then converted into a log-Mel spectrum and delivered through an encoder; the decoder anticipates the correct text caption, combined with specific tokens that steer the single model to accomplish numerous tasks, including multilingual speech transcription, speech translation to English, and timestamps at phrase-level.
Vodia announced a beta version of our PBX that connects a phone system to OpenAI realtime API (beta version) in November of 2024. We are delighted we can now provide our customers with a cloud pathway to leverage the power of Whisper AI.
Getting Started with OpenAI Cloud Transcription
To utilize OpenAI's cloud transcription, an OpenAI account and API key are required.
OpenAI Account: Navigate to the OpenAI platform and create or log in via Google, Microsoft, or email.
API Key Retrieval: Access the API Keys page, generate a new secret key, and securely copy it. This key is displayed only once.
Vodia Integration: Within tenant general settings, enable transcription and input the OpenAI API key.
Upon completion, all calls will be transcribed and available within the user portal.
Accessing Call Transcriptions
To view the transcribed content, simply log in to your user portal, navigate to the History section, select the desired call, and review the Call Content area.
In 2026, a modern phone system must go well beyond basic calling. Core requirements now include built-in AI for smarter call handling and transcription, real-time analytics dashboards for visibility and control, flexible auto attendants to route calls efficiently, seamless Microsoft Teams integration, and robust mobile apps that support hybrid and remote work. Clear separation between business and personal calls protects work-life balance, while reliable white-glove support ensures these capabilities work smoothly in real-world environments as communication needs evolve.
Legacy phone systems may still work, but they often come with hidden costs, limited scalability, and little support for hybrid work. Aging hardware, ongoing maintenance, and rigid infrastructure can quietly hold businesses back as they grow. Cloud-based VoIP systems remove these constraints by reducing telephony expenses, improving flexibility, and enabling teams to communicate seamlessly from anywhere. For many organizations, modernizing business telephony is no longer optional, it is a practical step toward efficiency and resilience.
A streamlined integration connects the Vodia PBX with the ElevenLabs Voice AI Platform using a lightweight IVR JavaScript script and native SIP REFER for call transfers. Audio and call control are handled entirely through standard SIP signaling, while all conversational logic, prompts, voice selection, and routing rules are configured in the ElevenLabs dashboard. This approach removes the need for webhooks or WebSocket connections and keeps the PBX side intentionally minimal, making the deployment clean and production-ready.
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