Hi
at the moment we have an asterisk box (1.4.7.1) running our telephone system. the * box has SIP trunks (voicepulse) which are suffering from quite bad jitter.
The asterisk server is currently in the UK, but we have offices in the states that use the SIP Trunk for incoming and outgoing calls.
the latency seems to happen as the packets cross the atlantic (tracked using “pingplotter”)
so my question is, if install an asterisk server in our USA premises an have that register the Voicepulse Trunk that will sort the latency out, but is it possible to get the two asterisk servers to be able to exchange calls i.e will the users in the USA be able to dial an extension in the UK?
sorry for the long post!
thanks