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!
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