On Wed, 2003-07-02 at 04:26, John Todd wrote: > You may be correct about the Via: header, but you're incorrect in the > concept as to how it relates to Asterisk, notably in your reversal of > what side of the transaction is putting data in the Via: header to > make SIP work correctly. > > This is cluttering up the list. Talk to me off line if you want a > better understanding of how NAT and SIP work with Cisco devices. > > Again, for those of you who might be trying to figure out what the > result of this conversation is: SIP clients behind NAT works fine in > both directions (incoming and outgoing calls), Asterisk makes it > work, it's not using STUN. Cisco devices work especially well. > > JT
Hi John, Thanks for the very helpful info so far. I concur with Richard Alexander's request to keep this discussion on list. How about Asterisk and NAT? Can you please comment if the examples below also work. 1x SIP phone <-> NAT box <-> Internet <-> NAT box <-> Asterisk 10x SIP phone <-> NAT box <-> Internet <-> NAT box <-> Asterisk The SIP phone(s) and Asterisk server are on private IP addresses. The NAT boxes (e.g. adsl router) have a public IP address. Any requirements for the NAT boxes like being a SIP proxy? Thanks, Patrick _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
