said by AndrewZ:said by brg:Question 2: Same as above; but assume inbound calling to one of my DIDs from a VoIP provider. I'm registered to that provider from my asterisk box. No opening/forwarding of SIP port(s) to the * box needed? Because of my registration?
No forwarding for SIP ports, because of your registration AND keepalives.
You will need to forward your RTP ports if you want to forward incoming calls through your server.
RTP is forwarded; thanks. A much smaller block than the huge block recommended as "standard." It's just me; not an office full of callers...said by AndrewZ:said by brg:Question 3: Assume inbound calling to one of my DIDs from a VoIP provider and I'm registered to that provider direct from my ATA -- no asterisk box. No opening/forwarding of SIP port(s) to the ATA needed? Because of my registration?
Correct, because of your registration AND keepalives.
