Add AI to the Phone System You Already Run
A conversational IVR, live transcription, call summaries, sentiment alerts and agent assist on Asterisk, FreeSWITCH or Kamailio, without replacing your numbers, queues or reporting.
Quick answer
What is AI VoIP integration?
AI VoIP integration adds artificial intelligence to a phone system you already run, instead of replacing it. Call audio is streamed from Asterisk, FreeSWITCH or a SIP edge such as Kamailio to speech-to-text, a language model and, where needed, text-to-speech. That enables a conversational IVR, live transcription, call summaries, sentiment and keyword alerts, smarter routing and real-time agent assist, while your numbers, queues and reporting stay as they are. When the AI should handle whole calls on its own, that is an AI voice agent.
What AI Adds to Your Calls
Each capability can be added on its own, and they share the same audio connection to your phone system.
Conversational AI IVR
Callers say what they need instead of pressing keys. The AI understands the request, asks a follow-up if needed and routes or resolves the call, with a keypad fallback for callers who prefer it.
Live call transcription
Real-time transcripts of calls from your PBX, stored against each call record, searchable, and available to supervisors, compliance teams and analytics.
Call summaries and QA
An automatic summary, reason and outcome for every call, written to your CRM or helpdesk, plus quality checks against your scripts and policies.
Sentiment and keyword alerts
Live alerts when a caller is frustrated, a competitor is mentioned or a required phrase is missed, so supervisors can step in during the call.
Smarter routing
Route each call on what the caller actually says, plus CRM data and queue state, instead of menu choices alone.
Real-time agent assist
Suggested answers, relevant knowledge-base articles and next steps appear in the agent's screen while the caller is talking.
Multilingual calls
Detect the caller's language and transcribe, summarise or respond in it, so one phone line serves several languages.
Fraud and deepfake checks
Voice biometrics and synthetic-voice detection for high-risk calls, with anomaly checks on calling patterns.
AI Integration or an AI Voice Agent?
Both use the same building blocks. The difference is who handles the call.
| AI VoIP integration (this page) | AI voice agent | |
|---|---|---|
| Who handles the call | Your staff or your existing IVR, helped by AI | The AI, end to end, with handover to staff |
| Typical uses | Transcription, summaries, QA, alerts, routing, agent assist, smarter IVR | Answering, booking, order status, reminders, outbound calls |
| Risk if the AI is wrong | Low: a person is still on the call | Higher: needs guardrails and easy handover |
| Good first step | Transcription and summaries on every call | One routine call type, then expand |
For AI that handles whole calls, see AI voice agent development and our case study of an AI voice agent added to a contact centre's FreeSWITCH platform.
Where AI Connects to Your Phone System
Every major open-source telephony platform has a standard way to stream call audio out and control the call. Choosing the right one is most of the integration work.
| Platform | How AI connects | Guide |
|---|---|---|
| Asterisk | AudioSocket or ARI external media streams call audio to your AI service; AGI or ARI controls the call (transfer, play, hang up) | Read more |
| FreeSWITCH | mod_audio_stream sends call audio over WebSocket; the Event Socket (ESL) controls the call and reads events | Read more |
| SIP trunk to an AI platform | The PBX sends selected calls over SIP to an AI platform such as LiveKit, and takes them back for handover | Read more |
| Kamailio or OpenSIPS edge | The SIP edge decides which calls go to AI services and which go straight to the PBX | Read more |
How It Works
From call audio to action, with each stage streaming to the next.
Tap the audio
Call audio is streamed from the PBX (AudioSocket, ARI external media or mod_audio_stream) to the AI service, usually over a WebSocket.
Speech-to-text
A streaming speech-to-text service, hosted or self-hosted, turns the audio into text as the caller speaks.
Language model
A language model classifies intent, writes summaries, spots keywords or decides the next action, using your knowledge base and rules.
Speak, if needed
For a conversational IVR, text-to-speech plays the reply back into the call. Transcription and analytics stay silent.
Act on the call
The PBX carries out the action: transfer to a queue, play a message, create a ticket or update the CRM.
Store and report
Transcripts, summaries, intents and scores are stored for search, QA and dashboards, with retention rules you set.
What Makes AI VoIP Integration Work in Production
AI VoIP integration sits between two specialisms: AI application development and telephony engineering. Many AI developers have never worked with SIP, RTP, Asterisk or FreeSWITCH, and many VoIP engineers have not built language-model applications. Projects go well when one team understands both sides.
Latency where it matters
A conversational IVR has to reply quickly, ideally starting about a second after the caller stops speaking. That needs streaming speech-to-text, a fast language model that streams its answer, streaming text-to-speech and services hosted close to the PBX. Transcription, summaries and analytics can run a little behind the call without anyone noticing.
Keeping call data under control
Calls contain personal data, and many organisations have rules about where it may go. Hosted speech and model services send audio or text to the provider. Where that is not acceptable, the same capabilities can run on your own servers: self-hosted Whisper for speech-to-text, open-weight language models served with vLLM, and local text-to-speech. Your compliance team confirms the final set-up.
Start small, measure, expand
The safest first step is usually transcription and summaries on every call: they help immediately, carry little risk and give you the data to decide where a conversational IVR or a voice agent will pay off. See our guides to replacing an IVR with AI and testing and monitoring voice AI.
AI VoIP Integration FAQ
What is AI VoIP integration?
Adding AI to an existing VoIP or PBX system: streaming call audio to speech-to-text and a language model to provide a conversational IVR, live transcription, call summaries, sentiment and keyword alerts, smarter routing and agent assist. The phone system stays in charge of calls; the AI adds understanding.
Does it work with our existing Asterisk or FreeSWITCH system?
Yes. Asterisk exposes call audio through AudioSocket or ARI external media and call control through AGI or ARI. FreeSWITCH offers mod_audio_stream for audio and the Event Socket for control. The AI is added as a layer; your numbers, routing, queues and call records stay as they are.
What is the difference between AI VoIP integration and an AI voice agent?
Integration adds AI to calls that your system and staff still handle: transcription, summaries, alerts, routing and agent assist. An AI voice agent answers or makes calls on its own and holds the whole conversation. Many projects start with integration and add a voice agent for routine calls later.
Which speech and language models do you use?
We choose per project and test on your own calls: hosted speech-to-text such as Deepgram or Google, or self-hosted Whisper; language models from OpenAI, Anthropic or Google, or open-weight models on your own servers. Self-hosting keeps call audio and transcripts inside your infrastructure.
How fast does it need to be?
Transcription, summaries and analytics are not time-critical. A conversational IVR is: the reply should start about a second after the caller stops speaking, which needs streaming at every stage and services hosted close to your PBX.
What does it cost?
It depends on which capabilities you add and how many calls you handle, so we scope it after a short discovery call. For AI voice agents that handle calls on their own, our guide ranges are $8,000–$20,000 for a managed-platform MVP and $25,000–$60,000 for a custom production agent. Speech and model usage are separate running costs.
Flexible Engagement
Need a Dedicated AI VoIP Developer?
Add an engineer who knows both telephony and AI to your team. Profiles in 1–2 days, a start within a week, and you own the code.
Ready to Add AI to Your Phone System?
Tell us which platform you run and what you want AI to do. We'll reply within one business day with a recommended approach.