[Asterisk-Users] More NAT questions

[Wilson Pickett]

> The rtp ports used for voice (10000:20000 in your example) vary by phone type.
> Cisco uses a different range of ports, Xten another range, Grandsteam yet
> another. The ports you have listed are what asterisk uses and are probably
> not the same ports as what your remote phones use. Therefore, the exact ports
> that you need to open are dependent upon exactly which phones you deploy,
> and on well you understand the handshaking that goes on end-to-end when
> establishing a sip call.

Yes. And if you want to stay with asterisk at 10000, you can tell
Grandstream and X-Lite to use those, I have no experience with the
others. I use this and port forwarding to go between two locations,
both of which have Linksys consumer NAT routers.