Integrators · SIP trunking
Carrier SIP trunks for systems your customers already run.
VoiceTel SIP Trunking connects PBX, UC, contact-center, and SBC environments to VoiceTel for PSTN origination, termination, numbers, emergency-service support, and route controls, without replacing the customer's existing call platform.
Deployments
Digest or IP authentication, Anycast or Geocast access, on the platforms your customers run.
Built for practical SIP deployments including Asterisk, FreePBX, FreeSWITCH, FusionPBX, SBCs, hosted PBX platforms, and contact centers. VoiceTel supports Digest and IP authentication methods, along with Anycast and Geocast network access.
- Digest authentication Supported
- IP authentication Supported
- Anycast Supported
- Geocast Supported
- Customer-owned SBC Supported
Why integrators choose VoiceTel SIP Trunking
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Deploy with what customers already have
Connect existing PBX, UC, SBC, and contact-center environments without forcing a full platform replacement.
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Registration or IP-auth designs
Choose the SIP authentication method that fits the customer architecture.
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Redundancy and rollback
Primary trunks, failover trunks, POP preference, number routing, and cutover rollback before migration.
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Emergency calling support
Tie emergency-service configuration to numbers, service addresses, and clear limitations.
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Bring your own SBC
Use customer-owned SBCs when architecture, security, or compliance requirements call for it.
SDN appliance
Solve difficult customer networks with the SDN appliance.
Many SIP deployments fail at the customer edge: NAT behavior, SIP ALG, UDP timers, firewall restrictions, one-way audio, registration instability, or RTP pinhole issues. The VoiceTel SDN appliance sits inside the customer LAN and establishes a secure tunnel to VoiceTel, so PBX registrations and media flows avoid them, with fewer multi-day troubleshooting cycles for you.
Configure your first SIP trunk.
Open a partner account, choose registration or IP-auth, and route a test call.
