Documention on SIP trunk parameters

For the life of me I can’t find any documentation on the parameters of creating a SIP trunk. I can find several examples of setting up trunks – all differing – but no real explanation of why and how certain parameters are being used. In my case, I’m trying to get freepbx to work with a VOIP-PSTN gateway, and none of the examples I’ve seen work, and I need to know what parameters canreinvite, context, qualify, type, etc. can take so I can modify them to work with my system. Are there any links to a listing of parameter definitions?

This should get you started:

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

HTH.

Thanks!