Hi,
My SIP service provider terminates calls in meetme in my Asterisk PBX
and am getting delay on those channels. I found following link to
measure delay in meetme and to decrease it eventually.
http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html
It says, enable USE_RTC for dahdi_dummy.
I have been using virtual server for hosting Asterisk and I had it
disabled as per one had mentioned here to prevent the crashing which
was happening
earlier..(http://www.odindev.com/content/troubles-zaptel-centos-52-xen).
Can you shed some light on the issue?
--SM
