Connect your PBX, contact center, or voice app to the world with elastic SIP trunks, global DIDs, HD audio, and carrier-grade reliability.
100+
Countries
HD
Voice Quality
99.99%
Uptime SLA
24/7
Support

Global DIDs
Instant provision
Elastic
Capacity
Failover
Always online
Everything you need to connect enterprise voice systems with reliability, clarity, and scale.
Provision local and international DIDs across 100+ countries with instant activation and flexible number inventory.
Full-duplex SIP trunking for inbound and outbound calling with transparent, per-minute pricing.
Crystal-clear audio with G.711, G.729, Opus, and wideband HD codecs optimized for enterprise quality.
Multi-path routing and redundant PoPs keep calls connected even during carrier or network outages.
Scale concurrent channels on demand—from a handful of lines to thousands—without hardware upgrades.
Port existing numbers with guided LNP workflows, status tracking, and minimal downtime.
Optional dual-channel recording with secure storage, retention policies, and compliance-ready export.
Global points of presence reduce latency and improve call quality for regional and international traffic.
Multi-region infrastructure designed for enterprise voice workloads.
Low-latency edge points of presence
Multi-carrier failover routing
TLS, SRTP, and IP ACL options
Quality metrics and call analytics
From contact centers to programmable voice platforms.
Power inbound queues and outbound campaigns with elastic SIP capacity and high concurrent call volumes.
Connect on-prem or cloud PBX systems to the PSTN with standard SIP and secure trunk authentication.
Handle seasonal spikes with elastic channels, failover routes, and multi-region termination.
Reach customers worldwide with competitive international rates and local DID presence.
Terminate and originate calls for programmable voice platforms, IVR, and notification systems.
Keep critical voice paths online with redundant trunks, overflow routing, and automatic failover.
Track concurrent sessions, ASR, ACD, MOS, and latency across every trunk with live dashboards and alerts.
Transparent usage pricing with volume discounts as you scale.
Pay-as-you-go SIP for small teams
Best value for growing call volumes
Dedicated capacity and global coverage
Point your PBX at our SIP endpoints or manage numbers and trunks programmatically with the Sendexa REST API.
// Provision DIDs with sendexa-sip
import { SipClient } from 'sendexa-sip';
const sip = new SipClient({
apiKey: process.env.SENDEXA_API_KEY,
});
// Search available numbers
const available = await sip.numbers.search({
country: 'GH',
type: 'local',
limit: 5,
});
// Purchase a DID and attach to trunk
const number = await sip.numbers.purchase({
phoneNumber: available[0].phoneNumber,
trunkId: 'trunk_primary_us',
});
// List active DIDs via REST
// GET /v1/sip/numbers
console.log('Provisioned:', number.e164);100+
Countries
50M+
Minutes / mo
5,000+
Businesses
99.99%
Uptime
Everything you need to know about Sendexa SIP Trunking.
SIP trunking connects your PBX or voice application to the public telephone network over the internet using Session Initiation Protocol. You get inbound and outbound calling without traditional PRI or analog lines.
Calls are billed per minute based on destination and direction. DID numbers are billed monthly per number. Business and Enterprise plans unlock lower per-minute rates and bulk DID packages.
We support G.711 (PCMU/PCMA), G.729, Opus, and HD wideband codecs. Codec preference can be negotiated per trunk for quality and bandwidth balance.
Yes. Number porting is supported in most regions. Our team guides you through LOA, carrier validation, and cutover scheduling to minimize downtime.
You can configure primary and secondary SIP endpoints, overflow numbers, and multi-region routes. If a path fails health checks, traffic is automatically redirected.
Yes. Optional dual-channel recording can be enabled per trunk or call. Recordings are stored securely with configurable retention and export options.
Use the Sendexa dashboard or REST API to search, purchase, and manage DIDs. Concurrent channel limits can be adjusted instantly on Business and Enterprise plans.
Yes. New accounts can trial SIP connectivity with test credits, sample DIDs where available, and sandbox endpoints so you can validate call quality before going live.
Connect SIP trunks, provision DIDs, and scale concurrent channels with carrier-grade reliability.