Adding a SIP line instead of a SIP Trunk

I’ve been provided with SIP account credentials, server, username and secret only. How do I set this up in FreePBX?

The same way you would set a trunk, but it is possible that your provider is not expecting to see the call coming from an Asterisk server, but from an endpoint, like an ATA, a desk phone or a soft phone.

Many single line providers block Asterisk as a “phone”. There’s a field in the SIP header that tells the other end what the agent type is.

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