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SIP Trunking

SIP Trunking for Contact Centers & Enterprises

Concurrent-channel SIP trunks with elastic capacity, DID options and codec flexibility for PBX and dialer platforms.

What you get

  • Direct and Tier-1 carrier paths
  • Live ASR / ACD / PDD monitoring
  • Proactive route swaps before quality drops
  • Prepaid or postpaid billing, USD / USDT / bank wire
  • 24/7 NOC on WhatsApp, Telegram and email
  • Test routes within 10 minutes on approved accounts

Typical specifications

ASR45–60% (destination dependent)
ACD3–6 min
PDD< 4 s
CLIAvailable on request
CodecsG.729, G.711 a/u, Opus
ProtocolsSIP (UDP/TCP), H.323 on request
Billing1/1 or 60/60
SettlementUSD · USDT · Bank wire

Popular destinations

Need a destination not listed? We quote any A-Z route.

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Frequently asked questions

Which PBX and dialers are supported?

Any SIP-compliant platform: Asterisk, FreePBX, VICIdial, 3CX, Vicibox, GoAutoDial, cloud dialers and more.

Which codecs?

G.711 a/u, G.729 and Opus.

How is it priced?

Per concurrent channel plus per-minute termination, or bundled minutes for predictable billing.

Is there failover?

Yes — redundant SIP proxies and automatic route failover.

Ready to test SIP Trunking?

Send one destination and your daily minutes — test routes in 10 minutes on approved accounts.

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