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Hello, I’m having trouble figuring out how to setup Asterisk
so that it’s only a registrar – not passing any RTP data during
phone calls. So far I got this far: Asterisk server holds registration information for phones Phones register with Asterisk giving it their ip+port where
they can be currently contacted NAT doesn’t seem to be a problem because STUN seems to
take care of it nicely for me. The hard part that I don’t understand is this: Phones can call each other BUT all the RTP traffic is passed
through Asterisk … I don’t want this, I need that the phones call
each other directly based on the registration info stored in Asterisk. I’m
having hard time wrapping my head around this – I think I’m missing
some key part – but the way I understand Asterisk is that it listens for
requests on the SIP channel, when it gets a request it handles it appropriately
using it’s dial plan. But in the dial plan the only thing that
makes sense to use is “dial” and once I do that all the RTP is sent
through asterisk (in-out) to the other phone… right? Or maybe the problem is on the phone setup? I tried to
make sure that I’m not specifying any outbound proxy but I do have to
specify “proxy” otherwise it will not know where to register …
right? Or maybe I’m all messed up 8-P … I thought I
understood asterisk at least a *bit*
until I came across this :-) Thanks for any clarification, Tomas |
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