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.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
