Keeping * out of the loop


#1

Need a little insight here.

If I have two sip phones on a network, and a session is initiated, I’m assuming * backs out of the call once it is established. Is this the case, and if not, what do I need to do to do that, and what are the ramifications? I’m trying to reduce cpu load here (assume both phones have compatible codecs)

What is the effect of ‘allow reinvites’ ?

rgds


#2

First look here:

forums.digium.com/viewtopic.php?t=2351

voip-info.org/wiki/view/Aste … media+path

voip-info.org/tiki-pagehisto … t+directly

The codec translation is no the only one reason for asterisk to stay in the media-path.
C.