The phone network still terminates most business conversations, so TeleQuick Voice speaks it natively. A carrier delivers a call over SIP, the engine answers it as a back-to-back user agent (B2BUA), decodes the G.711 media, and republishes it as an audio track your AI agent — or a human — can talk over. Every leg keys off a single call_sid, so signalling, media, recording, and reporting all line up on one call. This page orients you to that bridge. Follow the links to the task pages when you are ready to wire a trunk, provision a number, or hand a call to a person.
There is no separate PBX to run. The engine embeds its own SIP gateway, registrar, and dialplan — a FreeSWITCH-style modular core — so migrating off Asterisk or FreeSWITCH is a matter of re-pointing your trunks and agents, not standing up new signalling infrastructure. See Migration.

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How a phone call reaches your agent

A call crosses four boundaries between the carrier and your agent. Each is a place you configure something.
1

Signalling — SIP gateway / B2BUA

The carrier sends an INVITE to your per-tenant SIP host, <workspace-id>.sip.telequick.dev. The SIP gateway authenticates it (source-IP ACL for trunks, digest auth for registered devices), matches it to a trunk, and answers it. Because the engine is a B2BUA, it owns both legs of the dialog — caller on one side, agent or callee on the other — which is what makes transfer, bridging, and handoff possible.
2

Routing — dialplan and ACD

Once answered, a decision engine resolves what happens next from the matched trunk and the dialed number: stream the call to an AI agent, run an IVR/queue program, or ring a human. Number-to-destination mapping and skill-queue routing live in the PBX / ACD layer.
3

Media — RTP plane

Audio flows as G.711 RTP (µ-law PT 0 / A-law PT 8). The in-engine media plane decodes it to the runtime’s 8 kHz PCM16 bus, applies noise suppression, scores call quality (MOS), and publishes the caller’s voice as the MoQT track voice/<sid>/uplink. The agent’s reply comes back on voice/<sid>/downlink and is paced out as G.711 every 20 ms. Codec details are in the Codec Guide.
4

Handling — agent or human

The AI runtime attaches to those tracks and runs your ASR/LLM/TTS or speech-to-speech pipeline (see Runtime). At any point you can hand the live call to a person — the AI leg pauses and a human subscribes to the same caller track. See AI-to-Human Handoff.

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The pieces you configure

SIP Trunking

Connect a carrier. Bring your own SIP trunk (matched by source IP) or register outbound, in either direction, with credentials sealed at rest.

Number Provisioning

Import a DID, assign it to a trunk and an agent, and route inbound calls off it. Each workspace also gets its own <workspace-id>.sip.telequick.dev host.

Inbound Calls

Receive calls from the phone network, resolve the trunk and number, and route to an AI agent, an IVR, or a queue.

Outbound Calls

Originate a single call or run a paced bulk campaign out over a trunk, with an AI agent or a playback app on the far end.

PBX / ACD

Route through VDNs, skills, and queues; ring human agents on browser softphones or real SIP deskphones.

AI-to-Human Handoff

Warm-transfer a live call from the AI agent to a person, with the caller audio bridged the whole way.

Call Lifecycle

The full signalling and media sequence — from INVITE and early media through teardown, CDR, and Q.850 cause.

Codec Guide

Opus, PCM16, and G.711 — what runs on the wire, what the runtime works in, and where transcoding happens.

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Trunks: the unit of connectivity

A trunk is how a carrier — or another PBX, or a vendor bridge — connects to your workspace. Inbound INVITEs are matched to a trunk by source IP or by the dialed DID; outbound calls select a trunk to leave over. Each trunk carries the SIP and RTP addressing for both sides, credentials (sealed), a media mode, and a realm.
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direction
inbound | outbound | both
Which way calls flow over this trunk.
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media_mode
proxy | direct
default:"proxy"
proxy routes RTP through the engine’s media plane (the default, and required for AI handling and recording). direct is defined for pass-through media but the live runtime currently degrades it to proxy.
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realm
internal | external | ai_vendor
Classifies the far side — an internal device domain, an external carrier, or a vendor bridge.
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inbound_rule / inbound_skill_id / vdn_id
string
What an inbound call on this trunk does: stream to an AI agent, drop into a skill queue, or enter a VDN program.
Trunks are created and edited from the control-plane console at agent.telequick.dev (or via the admin API). On save they are sealed and pushed to the engine’s live configuration store — they are not kept in your reporting database. Full field reference and setup steps are on the SIP Trunking page.

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Numbers and per-tenant SIP hosts

When your workspace is provisioned, TeleQuick creates DNS records for <workspace-id>.sip.telequick.dev (and a matching WebRTC host) pointing at the engine. The engine reads the leftmost label of an inbound INVITE to identify your tenant, so multiple workspaces share one edge with no cross-tenant collisions. You then attach phone numbers (E.164 DIDs) to a workspace and bind each to a trunk and an agent. Imported and manually configured numbers route calls today.
The in-console “buy a number” search catalog is a demo stub (India-only, placeholder vendor) and does not purchase real numbers yet. To go live, import a DID you already own from Twilio, Telnyx, Bandwidth, or another provider and point its trunk at your workspace. See Number Provisioning.

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Placing calls out

Originating is the mirror image of inbound. A control-plane call mints a session and asks the engine to dial: 
import { Voice } from "@telequick/sdk";

const voice = new Voice({ apiKey: process.env.TELEQUICK_API_KEY });

// Dial a number and drop it straight onto an AI agent.
const call = await voice.calls.originate({
  to: "+15551234567",
  from: "+18005550100",   // a DID on one of your outbound trunks
  app: "AI_BIDIRECTIONAL_STREAM",
  agentId: "agent_42",
});

console.log(call.callSid);

from telequick import Voice
import os

voice = Voice(api_key=os.environ["TELEQUICK_API_KEY"])

call = voice.calls.originate(
    to="+15551234567",
    from_="+18005550100",
    app="AI_BIDIRECTIONAL_STREAM",
    agent_id="agent_42",
)

print(call.call_sid)
For volume, a paced bulk dialer runs campaigns of up to tens of thousands of numbers with configurable calls-per-second and max-concurrency limits, and the engine owns the pacing. See Outbound Calls.

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Every call is observable

The SIP gateway emits a lifecycle event on each transition — initiated, established, ring, hold, transferred, cleared — and writes a CDR (with Q.850 cause and duration) when the call ends. The media plane writes per-call quality (loss, jitter, MOS) alongside it. A segment tagged with a handler shows whether that stretch of the call was AI-handled or human-handled. All of it lands in your reports. See Observability.

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Next steps

Connect a carrier

Set up your first SIP trunk, inbound or outbound.

Take your first inbound call

Route a DID to an AI agent end to end.

Understand the lifecycle

Walk the full INVITE-to-CDR sequence.

Hand off to a human

Bridge a live AI call to a person.