> 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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
