[Asterisk-Users] sip to sip echo with meetme, timing

[Jerry Bonner]

When calling from sip phone to sip phone ( cisco 7940 ) we have very little or 
no echo. When conferencing through meetme through a sip only server, we 
experience lots of echo.

Would this have anything to do with the timing 
source?

The server is using ztdummy on 2.4 with uhci usb. Would using 
digium hardware timing help with this? Or switching to 2.6?

first time post, thanks for your comments / suggestions...

~jerry