SIP Trunk Configuration

A SIP trunk should be simple: point your PBX at the carrier and make calls. In practice it is credentials, IP allow-lists, codec negotiation, NAT traversal and number formatting — any one of which will silently break calls. I get trunks connected and working, whichever carrier you use.

What's included

• Trunk configuration for any SIP carrier (registration or IP-authenticated)
• Inbound DID routing and outbound caller ID configuration
• Codec negotiation and E.164 / national number-format handling
• NAT traversal so media flows in both directions
• Failover and least-cost routing across multiple carriers
• Capacity planning and concurrent-call sizing
• End-to-end test calls, both directions, before handover

Problems this solves

• Trunk will not register — 403 Forbidden or repeated 401 challenges
• Outbound calls rejected while inbound works, or the reverse
• Calls connect but there is no audio (the classic NAT/RTP problem)
• 488 Not Acceptable Here — codec or SRTP mismatch with the carrier
• Carrier says the problem is at your end and you need to prove otherwise

How we work

1. 1
   Talk

   Tell me the problem or the goal. No charge, no obligation.

2. 2
   Scope

   You get a clear plan, a fixed price, and a realistic timeline.

3. 3
   Build

   I implement and test it, keeping you updated as it progresses.

4. 4
   Handover

   Working system, documentation, and support after go-live.

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