Configure a SIP trunk on FreePBX
On FreePBX you do not write files: you fill in three screens. Below is what goes in each field, with the same details you would use on plain Asterisk.
1. Create the trunk
Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip). In the General tab give the trunk a name, then move to the pjsip Settings tab.
- Username: the SIP account username from the portal
- Secret: the password of the same account
- SIP Server: sip.vohippo.com — SIP Server Port: 5060
- Authentication: Outbound — Registration: Send
- Context: from-trunk (the one FreePBX uses for inbound calls)
2. The advanced settings
Same tab, Advanced section: these are the entries that make the difference between a clean call and one with audio on one side only.
- Codecs: only alaw, ulaw and possibly g722
- DTMF Mode: RFC 4733
- Direct Media: No
- Rewrite Contact: Yes — Force rport: Yes — RTP Symmetric: Yes
3. Outbound route
Connectivity → Outbound Routes. The route decides which numbers leave through this trunk.
- Trunk Sequence: the trunk you just created
- Dial Patterns: empty prefix and a match on X. to send everything, or per-country patterns
- Route CID: your verified number, if you are not setting it in the portal already
4. Inbound route
Connectivity → Inbound Routes: this is where you decide where your purchased numbers ring.
- DID Number: the number as the platform delivers it
- Set Destination: an extension, a ring group or an IVR
- If you are unsure about the DID format, leave the field empty and use the route as a catch-all until you see it in the logs
Let's talk: the demo is given by a person, not a video
One of our operators walks you through the portal live and answers the questions that matter for your case. If you would rather write, we are on WhatsApp.
Signing up is free and with no commitment: we credit your account with trial credit so you can make your test calls.