SIP trunk UI mssing fields

It seems that the user interface for setting up a SIP trunk should be a lot clearer and complete.
Are there not some “standard” requirements for a SIP trunk that could be added to the SIP trunk setup screen to avoid having to guess at the settings that need to be entered by hand?
The connection of a SIP trunk to an Incoming route should be made foolproof and perfectly clear.
I have not been able to get this working with a simple setup of Asterisk attached to 2 SIP trunks. Outgoing calls work and incoming do not. This seems to be about the simplest FreePBX setup possible.
After having set up Asterisk with manual configuration, I am disappointed that this is not dead easy in FreePBX.

A) It should simply be a matter of entering the SIP host, username and password and perhaps checking off a few boxes and entering correct information already on the SIP trunk UI now.
How many carriers provide SIP without a host, username and password? These are not fields on the screen.
B) The UI should make it much clearer about what needs to be done to connect the SIP trunk to an incoming route.

Not sure I understand the issue. Every trunk is different. We supply a dialog box you can use any Asterisk chan_sip parameter and convenient pre-configured contexts. In the inbound route you supply the DID digits and the destination.

Are you using the from-trunk or a derivative context? Have you checked out our wiki documentation?

I defer to your experience about carriers; I have only dealt wih one.
However, I am surprised that one can register a SIP trunk without a host or username or password.
It appears from reading forum items that a lot of people have problems in this area and there are a few common settings for SIP that need to be specified in most cases (host, secret, username, allow-guest, fromdomain, canreinvite,etc.) and could be fields in the gui rather than having to be guessed and typed in as Asterisk configurations.
I did not see any pre-configured contexts that I could select.
It certainly did not connect automatically to any inbound route nor did it provide any dropdown where I could select a trunk from the incoming route or an incoming route from the trunk.
This is not intended as a support request but I appreciate your willingness to help set me straight. I will ask in the support forum.
This is intended to start a development discussion about ways to make the simple SIP trunk a bit easier to setup in the GUI for 80%-90% of the users and to make the link between trunks and incoming routes more fool-proof and transparent.
Thanks for your quick response.

You are thinking in terms of your own carrier. If I am registering to my gateway for example I don’t need username, secret or any of those parameters.

The pre-configured contexts are documented, no drop down box.

The bottom line is you need to read the documentation, you can’t just wing it. If you have programmed sip peers in Asterisk you should know what goes in the box.

With finite development resources I don’t think this is an area that needs attention compared to larger development priorities. Also since channel_sip is replaced in Asterisk 12 I am sure no more development cycles will go into this module.