Connect Your PBX to the World
SIP Trunking & ISDN PRI, Done Right.

A SIP trunk is a virtual telephone line delivered over your internet connection using Session Initiation Protocol. A traditional phone line is a physical copper circuit. SIP trunks are lower cost, more scalable (you can add capacity instantly), and support features like DID numbers and simultaneous call paths that are expensive or impossible on analog lines.

ISDN PRI (Primary Rate Interface) is a digital telephone circuit that provides 30 simultaneous call channels over a single connection. It is appropriate when: your PBX is older and does not support SIP natively, your call volume consistently requires 20+ simultaneous calls, or your location has unreliable internet but reliable ISDN infrastructure. For most new deployments, SIP is the better choice.

SIP trunks are typically sold as concurrent call channels — each channel handles one simultaneous call. We size your trunk based on your peak simultaneous call requirement plus 20% headroom. A 50‑person office typically needs 10–20 concurrent channels depending on call volume.

A DID (Direct Inward Dial) number is a telephone number that routes directly to a specific extension or department without going through a receptionist or IVR. For example, your finance manager can have a direct number that rings their extension from outside. DID numbers are allocated as part of your SIP trunk configuration.

Yes — number porting transfers your existing telephone numbers to your new SIP trunk provider. The process takes 2–4 weeks depending on your current provider. We coordinate the technical configuration required on the PBX to receive ported numbers.

We configure failover routes for every SIP trunk deployment — typically routing calls to mobile numbers or a secondary SIP trunk on a different ISP if the primary connection fails. We also test failover before handover to confirm it works under real conditions.

We work with all major SIP trunk providers operating in India. We are not tied to any single provider — we recommend based on your location, call volume, and required features.

Yes. SIP trunk providers typically offer competitive international rates, and outbound international calling is configured as part of your dial plan. We set up per‑extension or per‑group international call permissions to control cost and prevent misuse.

A SIP trunk connects your on‑premise PBX to the public telephone network. A hosted PBX replaces your on‑premise PBX with a cloud‑based system. Layerix specialises in on‑premise PBX with SIP trunk connectivity — giving you full control of your system without dependency on a cloud service.

Provisioning a new SIP trunk typically takes 3–7 business days from order confirmation. Configuration on the PBX takes 1 day. Number porting adds 2–4 weeks. We provide a complete timeline before work begins.

Yes — each site can have its own SIP trunk, or a central SIP trunk can be distributed to remote sites over your WAN. We design the trunking architecture based on your site connectivity and call volume per location.