[asterisk-users] IAX vs SIP trunks between Asterisk boxes

[Thomas Kenyon]

Brad Templeton wrote:

> 
> For SIP phone calling * box, relay to other * box and out to SIP
> phone, you definitely want SIP all the way.
> 
Unless bandwidth between the * servers is a concern, then you're better 
off keeping the link between servers as IAX. (preferably trunked)

It is worth remembering in this sort of setup, often the phones at one 
site will not have a route to the phons on the other site, so the calls 
wont be re-invited off to the handsets anyway.