- SIP
- Call centers
SIP trunk setup checklist for call centers
A step-by-step checklist for connecting a call-center dialler or softswitch to a carrier SIP trunk: authentication, codecs, caller ID, capacity and first-call testing.
RazeConnect Team
3 min read
Most failed trunk launches come down to a missed setting rather than a bad carrier. A mismatched codec, the wrong authentication method or a caller ID in the wrong format can make a healthy route look broken. Work through this checklist before sending production traffic and your first day will be uneventful — which is exactly what you want.
Pick the right authentication
If your dialler sits on a server with a fixed public IP, use IP authentication: set the SIP account host to that address and the switch accepts calls only from it. If agents are distributed or behind changing IPs, keep the host dynamic and register with username and password. Never share one credential across sites; create one SIP account per location so you can trace and cut off traffic individually.
Match codecs and transport
- Offer G.711 first for best quality where bandwidth allows
- Enable G.729 for bandwidth-constrained sites
- Use UDP 5060 by default, TCP for large SIP messages, TLS 5061 when signalling must be encrypted
- Keep RTP ports open on your firewall for the media range
Set caller ID correctly
Send caller ID in E.164 format. Set a sensible default on each SIP account from the portal, then override per call from your dialler where your number plan requires it. Present only numbers you own or are authorised to use.
Size capacity and pacing
Predictive diallers can generate bursts of calls per second far above what your agents can handle. Agree CPS and channel limits with your carrier, configure your dialler to respect them, and watch the live calls view during the first ramp-up.
Protect against fraud
- Use long random SIP passwords and rotate them if staff leave
- Prefer IP authentication for servers
- Keep a prepaid balance that covers days, not months
- Set a low-balance alert
Place your first test calls
Call a mobile and a landline in your main destination, confirm two-way audio, check the caller ID shown on the handset and verify the call appears in your call history with the expected cost. Only then ramp up.
Questions about this guide? Talk to our routing team.