Hello
After much reading I have come to the realization that after years of
using Asterisk I know very little about Sip.
I have my Kamailio box up, I have Asterisk 1.8.x running with realtime
working. I thought it would be just a case of registering SIP trunks
from my provider to the kamailio and registering our internal asterisk
servers to the kamailio.
Much of what I read talks about using Asterisk as the PSTN interface,
but that interface is through a sip trunk purchased from a provider.
Won't Kamailio be the PSTN gateway? The idea here is to pool all the
sip trunks from the various hosted asterisk solutions (VM running
asterisk) and point them all to a proxy to facilitate the aggregation
of traffic.
I have been reading SIP tutorials and the mailing list archives. If
anyone has a sample config and perhaps a little direction it would be
highly appreciated.
Thank you
Greg
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