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You can refer to PrivateServer management documentation for creating a proper SIP Trunk. We are going to explain here which specific values need to be set up in order to create it properly.
In the fig. X you can watch a plain example of a working configuration. It's major differences from a standard one is that:
- check out the Register option
- Dtmfmode value must be RFC2833
- Allow codec is ulaw only
- Disallow codec must be all
The Virtual phone number is optional and can be left blank. After you're done with the Inbound you have to