Can anyone here help me understand what I missing with this setup. I want to use Asterisk as a feature server only, speaking only SIP (no IAX), and use SER for registration to minimize necessary bandwidth.

 

SIP-phone <-->SER <--> * <--> PSTN Provider <--> Regular-phone
Regular-phone <--> PSTN Provider <--> SER <--> * <--> SIP-phone

 

I want to allow SIP users to transfer calls to other users, either on the system or on the PSTN. I'm not sure how to make this work with *. From what I understand, once a call is setup by SER the caller has no access to * because * is not in the media path. If so, * would not be able to catch the DTMF tones and transfer the call. Is this correct?

 

Any help would be greatly appreciated!