How to test a new carrier trunk before going live

Before you point a live campaign at a new Carrier, add the Trunk in a staged state, confirm the SIP peer is reachable, then place exactly one manual test call. If the call connects with clean two-way audio and the right caller ID, you are ready to go live. If it does not, you fix it without a single real lead in flight.

Stage the trunk before you activate it

Add the carrier under Admin then Carriers then Add New Carrier, but set Active to N first. While Active is N, VICIdial leaves the trunk out of the auto-generated Conf file set, so it cannot place or take calls yet. That gives you a quiet window to double-check the registration string, the account block, the Codec line, and the dial pattern before anything touches the wire.

• Carrier ID and Carrier Name read clearly and have no spaces in the ID.
• Registration string or host line matches what the carrier gave you.
• Dialplan strip and prefix match the format the carrier expects, for example a leading 1 for North American numbers.

When the entry looks right, flip Active to Y and save. Wait about a minute for the conf files to regenerate. The trunk is only real once that rebuild completes.

Confirm the peer, then place one test call

On the Asterisk CLI, run sip show peers (or pjsip show endpoints for PJSIP). A registered or reachable peer means the trunk is up at the network level. If the peer is unreachable, stop here and fix registration or IP authorization before you waste a test call.

flowchart LR
  A[Add carrier Active=N] --> B[Flip Active=Y]
  B --> C[Wait ~1 min for conf rebuild]
  C --> D[sip show peers reachable?]
  D -->|No| E[Fix register or IP auth]
  D -->|Yes| F[Place one test call]
  F --> G[Two-way audio + caller ID?]
  G -->|No| E
  G -->|Yes| H[Go live]

Once the peer is reachable, place a single manual call from the Campaign Detail screen. Dial a number you control, like your own cell, so you can answer it and listen. You are checking three things at once.

1. The call actually connects through the carrier and rings the destination.
2. Audio flows both directions with no one-way silence or clipping.
3. The caller ID that lands on the receiving phone is the number you expect.

Go live only after a clean call

Only point production campaigns at the trunk once a manual test call connects with clean audio and correct caller ID. This same staging habit is worth repeating any time you add or change a carrier, which is covered across the VICIdial carrier integration guide. If you are adding the entry itself for the first time, walk through adding a VICIdial carrier trunk first, then come back to this test pass.

On a VICIfast box your server already has a fixed public IP and a clean Asterisk install, so the only variable in your test is the carrier itself. You get a dedicated dialer over HTTPS in under 40 seconds and can start staging trunks the same day.