Make your first outbound PSTN call

Two paths turn a new Orbit account into a phone that dials the PSTN:

Option A — register a SIP trunk with your own PBX or softphone software, and let API-origin calls dial out over it.
Option B — the browser softphone, where an agent dials straight from a web page with no SIP client to install.

Both end at the same place: an outbound PSTN call with a readable lifecycle (ringing → answered → completed). This guide walks either path end to end and links the deep-dive page for the path you pick.

Every outbound leg terminates through the Devotel softswitch. A trunk you register here is one origin surface; the termination path stays on Orbit’s own network either way.

1. Prerequisites

Which prerequisite applies depends on the path:

Option A (SIP/trunk): a SIP-capable PBX (FreeSWITCH, Asterisk, 3CX) or a desktop SIP client (for example MicroSIP or Linphone), plus inbound SIP-credentials you mint on the account.
Option B (browser softphone): an HTTPS page and a browser with a microphone grant — the same rules the browser softphone guide lists in full.
Either path: at least one voice-enabled DID in the account to present as caller ID, and an API key with voice scope if you dial via the API.

2. Buy a voice-capable number

If the account has no DID yet, work through Buy and provision numbers — search inventory with capabilities=voice (or sms,voice when messaging will share the number), purchase, and poll the order until the line reports provisioned:

curl "https://api.orbit.devotel.io/api/v1/numbers/available?country=US&type=local&capabilities=voice" \
  -H "X-API-Key: dv_live_sk_..."

Keep the E.164 of the purchased number — it is the from on every dial below, and it is the only caller ID the outbound surface accepts.

3. Option A — register a SIP trunk

This path connects your own phone system to Orbit. The full lifecycle (direction choice, probe-before-save, credential capture, routing, monitoring) is a nine-step walkthrough in Connect a SIP trunk; the short version for a first call:

Create the trunk with the carrier-facing details:

curl -X POST https://api.orbit.devotel.io/api/v1/voice/sip-trunks \
  -H "X-API-Key: dv_live_sk_..." \
  -H "Content-Type: application/json" \
  -d '{
    "name": "HQ PBX",
    "host": "pbx.example.com",
    "port": 5060,
    "transport": "udp"
  }'

Register the device that will originate calls (your PBX, or a SIP client on an agent’s desk) against the SIP edge host shown in Settings → SIP trunks, using the digest or IP-allowlist auth the trunk is configured with.
Skip to section 5 — once the originating surface exists, the dial step is identical to Option B.

A trunk that fails to register never carries the first call — run the pre-save probe (POST /api/v1/voice/sip-trunks/probe, exercised in sip-trunk-setup) and it names the failing phase (DNS, connect, or REGISTER) instead of a generic rejected call.

4. Option B — the browser softphone

The no-install path: mint a short-lived credential, connect the client, dial. The full guide is Set up the browser (WebRTC) softphone; on a first call you need three moves.

Mint a token:

curl -X POST https://api.orbit.devotel.io/api/v1/voice/softphone/token \
  -H "Authorization: Bearer dv_live_sk_..."

From your own page, mount the SDK softphone and connect — a minimal HTML page:

<!doctype html>
<button id="connect">Enable microphone</button>
<input id="to" placeholder="+14155550100" />
<button id="dial">Dial</button>

<script type="module">
  import { OrbitSoftphone } from "https://cdn.jsdelivr.net/npm/@devotel-orbit/web";

  const softphone = new OrbitSoftphone({
    apiBaseUrl: "https://api.orbit.devotel.io",
    apiKey: "dv_live_sk_...", // serve from your backend, never commit it
  });

  document.getElementById("connect").onclick = async () => {
    await navigator.mediaDevices.getUserMedia({ audio: true });
    await softphone.connect(); // mints token, publishes mic
  };

  document.getElementById("dial").onclick = () =>
    softphone.dial(document.getElementById("to").value);
</script>

Dial from the browser, or call the endpoint directly once the client is connected:

curl -X POST https://api.orbit.devotel.io/api/v1/voice/softphone/dial \
  -H "Authorization: Bearer dv_live_sk_..." \
  -H "Content-Type: application/json" \
  -d '{ "to": "+14155552671", "from": "+16572262362" }'

Credentials live 60 minutes — re-mint before expiresAt instead of recycling. The browser-fallback reality and the inbound decision (DNIS → queue or SIP routing for inbound rings) live in the softphone guide.

5. Dial a single PSTN call

With either origin surface registered, the dial step is one API call — POST /api/v1/voice/calls. Pass the bought DID as from and any PSTN destination as to; a record: true flag captures the audio for the post-call read:

curl -X POST https://api.orbit.devotel.io/api/v1/voice/calls \
  -H "X-API-Key: dv_live_sk_..." \
  -H "Content-Type: application/json" \
  -d '{
    "to": "+14155552671",
    "from": "+16572262362",
    "record": true
  }'

The 2xx response returns data.id (call_…) — the handle every later step uses. Options answer_url/staticVerbs, AMD, and metadata-level bookkeeping are at Program an outbound voice call; for a first call there is nothing to configure beyond to and from.

6. Read the lifecycle states

A call moves ringing → in_progress → a terminal completed on answer (or failed / no-answer on a declined or unanswered ring). Poll the record:

curl https://api.orbit.devotel.io/api/v1/voice/calls/call_9f2a1c \
  -H "X-API-Key: dv_live_sk_..."

…or subscribe to call.completed / call.failed webhooks under Settings → Webhooks and verify the delivery signature. The Voice calls page in the dashboard lists the same progression per call; the full state map and webhook payloads live in programmable-voice-call.

7. Common errors

These cover the failures a first outbound call hits in the wild:

Symptom	Cause	Fix
`401`/`407` on REGISTER, or trunk `unregistered`	SIP auth fails: wrong register username/password, or a digest challenge answered with a stale password	Re-check the credentials the carrier issued (Option A trunk) or the minted SIP credential (desk client). On Orbit-side incoming trunks, rotate the digest password — the plaintext is shown exactly once at create.
Call connects, then one-way or no audio	Codec mismatch or SRTP posture	On the trunk, drop to the codecs your carrier lists; on softphone credentials, SRTP is mandatory and an RTP-only client never establishes audio. See TLS/SRTP posture in sip-trunk-setup.
Outbound rejected right at originate	Outbound blocked	Emergency-shaped dials are rejected (`EMERGENCY_CALLING_NOT_SUPPORTED`). Otherwise the trunk’s routing rules (a destination blocklist) or the account’s spend cap and quiet-hours gate stopped it, or no trunk registered at all — the error returned on the dial names the gate.
`403 Organisation does not own phone number …`	`from`/`callerIdNumber` is not a DID the account owns	Buy or port the number first; outbound CLI can only be an owned DID.
Register probe never succeeds	DNS or firewall on the SIP edge	Probe before save (`POST /api/v1/voice/sip-trunks/probe`) names the failing phase explicitly.

SIP failures past this short list are covered in Troubleshooting: SIP trunk down or calls failing; lifecycle outcomes that look like (no-answer, busy, unreachable) map in programmable-voice-call.

Where to go next

Program every call — answer_url verbs, AMD, recordings, transcripts: Program an outbound voice call.
Run the trunk in production — failover, health snapshot, route quality: Connect a SIP trunk.
Give agents the full softphone — inbound routing, mobile push, SDK events: Browser (WebRTC) softphone.

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Make your first outbound PSTN call: SIP trunk or browser softphone

Make your first outbound PSTN call

1. Prerequisites

2. Buy a voice-capable number

3. Option A — register a SIP trunk

4. Option B — the browser softphone

5. Dial a single PSTN call

6. Read the lifecycle states

7. Common errors

Where to go next

​Make your first outbound PSTN call

​1. Prerequisites

​2. Buy a voice-capable number

​3. Option A — register a SIP trunk

​4. Option B — the browser softphone

​5. Dial a single PSTN call

​6. Read the lifecycle states

​7. Common errors

​Where to go next

Make your first outbound PSTN call

1. Prerequisites

2. Buy a voice-capable number

3. Option A — register a SIP trunk

4. Option B — the browser softphone

5. Dial a single PSTN call

6. Read the lifecycle states

7. Common errors

Where to go next