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<DIV style="BACKGROUND-COLOR: white">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.</DIV>
<DIV style="BACKGROUND-COLOR: white"> </DIV>
<DIV style="BACKGROUND-COLOR: white">SIP-phone <-->SER <--> *
<--> PSTN Provider <--> Regular-phone<BR>Regular-phone
<--> PSTN Provider <--> SER <--> * <-->
SIP-phone</DIV>
<DIV style="BACKGROUND-COLOR: white"> </DIV>
<DIV style="BACKGROUND-COLOR: white">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? </DIV>
<DIV style="BACKGROUND-COLOR: white"> </DIV>
<DIV style="BACKGROUND-COLOR: white">Any help would be greatly
appreciated!</DIV>
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