> ## Documentation Index > Fetch the complete documentation index at: https://docs.orbit.devotel.io/llms.txt > Use this file to discover all available pages before exploring further. # Jambonz softswitch model > How Orbit's voice softswitch surface (Jambonz) routes a call — SIP trunks, softphone registration, feature codes, inbound and outbound path selection, and the tenant-owned safety controls # Jambonz softswitch model Every PSTN call on Orbit — inbound DID traffic, outbound dialing, SIP trunk traffic from your own PBX, and desk-phone / softphone calling — terminates on the Jambonz session border controller. Jambonz is Orbit's voice softswitch edge: it handles SIP signalling, and asks Orbit for the routing decision and the verb plan for each call. This page walks through that surface the way a call sees it, so you can onboard SIP and predict which routing rule applies. ## Where the voice softswitch lives Three flavours of SIP endpoint all land on the same softswitch edge: * **Outbound SIP trunks** — the carrier connections you configure under **Settings → SIP trunks** (or via `GET /api/v1/voice/sip-trunks`) for outbound calling. Outbound PSTN termination exits only via the Devotel wholesale softswitch; a configured trunk that fails to register or stops answering falls back to that default path. * **Softphone / desk-phone registration** — SIP credentials you issue per extension (browser softphone, Yealink / Polycom / Grandstream desk phone). When a device sends REGISTER, the softswitch checks the digest against the credential you created, and stamps the last registration time. Manage credentials under **Settings → SIP credentials** or `POST /api/v1/voice/sip-credentials`. * **BYOC inbound trunks (Bring Your Own Carrier)** — calls originated from your own voice infrastructure (an Asterisk / FreeSWITCH / 3CX PBX, or an AI-voice stack you run yourself). These arrive on a shared inbound carrier and are authenticated by the digest credentials or IP allowlist you store on the trunk, then routed per the inbound route model below. ### Feature codes (star codes) A desk-phone user has no dashboard in front of them, so the handset can dial a star code — feature codes like `*78` for Do-Not-Disturb — and the softswitch resolves the dialed string to a typed action, speaks a short confirmation, and hangs up. Only codes the platform can honour end-to-end are recognised (Do-Not-Disturb on/off, call-park retrieve, voicemail check, malicious-call-id trace); everything else gets a polite "feature not available" prompt and a clean hangup. ## Inbound vs outbound route model The softswitch asks Orbit "what should I do with this call?" on every INVITE, and the answer differs by direction. ### Inbound (PSTN → your tenant) On an inbound INVITE the dialed E.164 number is resolved in this order: 1. **Exact match** — the `inbound_routes` row for the DID (voice queue, softphone ring list, IVR flow, voicemail, etc.). 2. **DNIS pattern fallback** — when no exact row exists, your `dnis_routes` rules are evaluated: prefix or regex patterns, ranked by priority, with the longest (most specific) pattern winning. Tenant scoping comes from the number's owning organisation, so a pattern you write can never catch another tenant's calls. 3. Once a route row is chosen, **business-hours and holiday calendars** on that row decide whether the primary destination or the configured fallback (closed-hours behaviour) applies. The softswitch then renders Orbit's answer as Jambonz "verb" sequences — `dial`, `say`, `enqueue`, `hangup` — and executes them for the life of the call. Ring strategies (simultaneous or sequential hunt across SIP credentials) are tracked per call so each no-answer step advances exactly once. ### Outbound (your tenant → PSTN) Outbound has two entry points: * **Registered devices** — an INVITE from a softphone / desk phone is validated against the credential: caller-id ownership, `outbound_enabled`, blocked countries, the concurrency caps described below, and a pre-flight billing hold. If approved, the call dispatches; otherwise it gets an accurate SIP response (403/402/480) and a clean teardown. * **API-initiated calls** — `POST /api/v1/voice/calls`. Both entry points terminate outbound PSTN traffic exclusively on the Devotel wholesale softswitch; a configured outbound trunk is an optimisation and a failover preference, never a third-party termination change. For BYOC inbound trunks, the guard chain runs before any verb is emitted: per-trunk routing rules (country/prefix blocklist, time-window check, caller-id rewrite), caller-id ownership against the trunk's allowed caller IDs or the org's owned DIDs, a fraud guard (rate and daily-spend caps, premium-rate prefix block), and a pre-flight billing hold. Failures answer the originating PBX with the correct SIP status, not a silent drop. ## SIP header conventions, registration caps, and concurrency * **Header lookups are case-insensitive.** Upstream proxies forward custom headers (`X-Orbit-Org-Id` and similar) with arbitrary casing, so every consumer of a SIP header matches names without regard to case. When you integrate your own SIP stack, you can send custom headers in any casing — they will still be found. * **Per-credential registration cap.** A SIP credential caps how many distinct endpoints may hold a live REGISTER binding at once. A keepalive REGISTER refreshes the endpoint's existing slot; only a genuinely new endpoint beyond the cap is rejected — so a stolen credential cannot be registered from a flood of devices. * **Per-credential concurrent-call cap.** Separately, each credential caps concurrent active calls, reserved atomically so two simultaneous outbound requests can't both slip under the limit. A stuck call that never reports its status frees its slot automatically after a bounded time-to-live rather than leaking the cap forever. Both caps are tenant-owned settings you set per credential; there is no platform-imposed ceiling beyond what you configure. ## Safety controls you own Per Orbit's compliance model, these controls are **yours to configure** — the platform provides them; you decide how strict to be: * **Abandoned-call safe harbor (US predictive dialing).** The dialer already keeps abandoned calls under the federal 3 % ceiling. For the calls that do get abandoned, you can configure a safe-harbor message — per campaign, with an organisation-level fallback for business identity — that plays within two seconds of the called party's greeting: your business name plus a callback number the party can use for a do-not-call request. Once configured, the message plays automatically on abandoned attempts before hangup. * **Caller-ID filter on inbound routes.** Each inbound route carries an optional caller-ID filter: block repeat spammers' numbers from ever entering a queue or ringing a softphone (the caller hears a polite "the call cannot be completed" message), or maintain a small VIP allowlist that rings through immediately. * **Caller-ID privacy.** Per-route caller-ID presentation rules decide what the called party sees (your DID, an anonymised identity, or a withheld presentation) where the destination carrier accepts it. ## Where to go next * Onboarding a trunk end-to-end: [SIP trunk connection](/guides/sip-trunk-connection) and [Registering your PBX on Orbit](/guides/registering-your-pbx-on-orbit). * DNIS pattern routing configuration: [DNIS pattern routing](/guides/dnis-pattern-routing). * When a trunk stops answering: [Troubleshooting: SIP trunk down or calls failing](/channels/voice/sip-trunks-troubleshooting). * How PSTN media crosses the edge: [Media planes](/concepts/media-planes).