Overview

What Is SIP Trunking?

Replace legacy ISDN/PRI lines with cost-effective SIP trunks. Connect your existing PBX, softswitch, or dialer to our voice network. Unlimited channels, pay per minute.

Replace Legacy Lines

Migrate from expensive ISDN/PRI circuits to flexible SIP trunks. Same reliability, fraction of the cost.

Connect Any PBX

Works with Asterisk, FreePBX, 3CX, Avaya, Cisco, Mitel, and any standards-based SIP PBX.

Unlimited Channels

No fixed channel limits. Scale concurrent calls based on your internet bandwidth and infrastructure.

Pay Per Minute

Transparent per-minute pricing with no monthly minimums. Volume discounts applied automatically.

Voice Infrastructure

Enterprise-Grade SIP Connectivity

Connect your PBX to premium A-Z voice routes with TLS/SRTP encryption. Redundant infrastructure across European data centers ensures five-nines reliability.

Key Features

Enterprise-Grade SIP Trunking

A-Z Global Termination

Premium voice routes to every destination worldwide. Direct carrier connections for the best quality and lowest latency.

Flexible Caller ID

Set custom caller ID per trunk, per call, or per destination. Present local numbers to improve answer rates.

Call Routing & Failover

Intelligent route selection with automatic failover to backup carriers. Ensure call completion even during outages.

Real-Time CDR Access

Full call detail records available instantly. Filter by date, destination, duration, and cost. Export or access via API.

Budget Caps & Rate Alerts

Set spending limits and receive alerts before reaching thresholds. Prevent bill shock with proactive budget controls.

TLS + SRTP Encryption

Secure signaling with TLS and media encryption with SRTP. Protect sensitive voice communications end-to-end.

Who It's For

Built for Voice-Heavy Operations

Contact Centers

High-volume outbound and inbound calling with predictive dialer integration and real-time monitoring.

VoIP Resellers

White-label SIP trunking for MSPs and telecom resellers. Competitive wholesale rates with flexible billing.

Businesses Replacing ISDN

Seamless migration from legacy ISDN/PRI to SIP. Keep your existing PBX and phone numbers.

Enterprise PBX Users

Connect your Asterisk, FreePBX, 3CX, or any SIP-compatible PBX to our carrier-grade voice network.

ISPs Offering Voice

Add voice services to your portfolio. Wholesale SIP trunking with full API management and white-label options.

Scalability

Flexible Capacity on Demand

Scale from 10 to 10,000 concurrent channels without renegotiating contracts. Add capacity in real time through the dashboard or API.

Getting Started

Start Making Calls in 4 Steps

Sign Up

Create your account and choose a SIP trunking plan. No commitment required — pay as you go.

Configure SIP Credentials

Set your SIP authentication credentials (IP-based or username/password) in your dashboard.

Point Your PBX

Configure your PBX to route outbound calls through our SIP proxy. We provide setup guides for popular platforms.

Start Making Calls

Your trunk is live. Make calls to any destination worldwide with real-time CDR tracking and quality monitoring.

Frequently Asked Questions

What codecs do you support?

We support G.711 alaw and ulaw (uncompressed HD audio), G.729 (compressed, bandwidth-efficient), Opus (adaptive quality), and G.722 (wideband audio). Codec negotiation is automatic during call setup — your PBX and our platform will agree on the best available codec. We recommend G.711 for LAN/high-bandwidth environments and G.729 for WAN/limited bandwidth scenarios. T.38 for fax over IP is also supported.

Ready to Get Started?

Create your account in minutes and start sending SMS or making calls today.