SIP dial issue

I’m trying to replace an old box with an asterisk 1.4 and freepbx 2.3 to a new asterisk 1.8 and freepbx 2.9.
My problem is that in the old box I have some custom extensions that I cant use in the new one.
These extensions are mappings to external numbers and have a custom dial like SIP/123456@ABCD, where ABCD is the trunk name of my voip provider that is created on freepbx.
In the new one this doesn’t work, I investigated and it seems asterisk 1.8 has some differences in the dial options,the @part has to be a hostname or ip of the provider.
But putting only the host/ip, the sip invite is refused because its not authenticated…
Is a way to pass also the peer authentication in the dial string?
Or is there other way to make this mappings?

Any help is appreciated, because I’m stuck and without ideas to solve this.