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
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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