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Claimable Voice Agents: Provisioned by code, claimed by teams

How developers can programmatically instantiate live phone agents that non-technical operators can manage.

Voice AI Team
Voice AI Team—Core Platform
SEP 10, 2026
Answrable
provision / claim
SIP

An agent can provision a voice backend, a human can claim it later

Claimable Agents implements programmatic SIP endpoint allocation and prompt graph provisioning via simple REST calls, allowing developers to spin up ephemeral voice lines that team leads can later claim and audit in the web dashboard.

Agents are the ones building now. They need voice primitives.

Until now, building an AI voice agent required cobbling together disparate vendors: a speech-to-text transcriber, an external LLM completion API, a text-to-speech voice synthesizer, and a legacy SIP trunk aggregator.

This chained architecture created 2,000ms+ roundtrip latencies, robotic speech turn jitter, and clipped conversations whenever callers tried to interrupt. By engineering our own 8-billion parameter foundation model (Answrable HCM-1), we achieved native full-duplex speech-to-speech comprehension with sub-180ms reflexes.

agent-workflow.tsAnswrable HCM-1 8B
import { AnswrableAgent, tool } from "@answrable/core";

export const clinicReceptionist = new AnswrableAgent({
  model: "answrable-hcm-1-8b",
  temperature: 0.2,
  turnReflexLatencyMs: 180,
  systemPrompt: "You are the autonomous voice receptionist for CareClinic...",
  tools: [
    tool({
      name: "verifyInsuranceAndBookSlot",
      description: "Checks real-time calendar availability and schedules patient",
      parameters: { patientId: "string", desiredTime: "string" },
      execute: async ({ patientId, desiredTime }) => {
        return await db.appointments.create({ patientId, slot: desiredTime });
      }
    })
  ]
});

What's included in Answrable GA

HCM-1 Foundation Model

8B parameter speech-to-speech weights with human-grade turn taking.

Direct Carrier SIP

Connect your own carrier numbers with zero transcoding overhead.

With full-duplex interruption handling, callers can speak naturally, pause to think, or change topics mid-sentence without robotic clipping.

What's next for voice telephony

We are expanding our regional edge points of presence across North America, Europe, and Asia Pacific to guarantee sub-50ms carrier jitter worldwide.

In the meantime, we want to see what you build with these tools. Tag us on X, send us feedback, and we're in Discord too.
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