you mean IAX isn't a standard :) Also IAX requires your call router / billing gateway to handle the voice traffic too (or you put your CDR recording at the end points) With SIP, just the signaling is needed, allowing more scalability. I recall talking about this at astericon but it never eventuated to anything. The idea was basically keeping the original channel open even on a native transfer.

-Adam

Tom Samplonius wrote:
On Mon, 14 Mar 2005 16:47:21 -0700, Joseph <[EMAIL PROTECTED]> wrote:

* SIP isn't a standard.  It could be made into an official standard,
if there was a standards document.  Someone should write one, and
start an IETF working-group.  If the IETF adopted it, there would be
wider acceptance.

* SIP NAT traversal in Asterisk is harder than it needs to be.  This
should be getting better.  But SIP in general isn't very easy to
configure in Asterisk.  It sounds like this is getting a lot better in
the next release (no more goofy "peer" vs. "friend" distinction).

Tom
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