hello friends,
iam trying for the sip and h323 conversion
with asterisk as the gateway .
does any body got success before in this
if so please guide me
in my intial configuration i am forwarding the calls
to the astersik ip
if we do the forwarding does we get the register
,invite etc methods .
does the call will go through the ser ip during the
entire call or else it directly go to the asterisk
and cutt of communication with the ser.
so guide me please in this regards
with regards
serdiehard
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