Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP capacity and IP bandwith and no codec conversion is needed - do you think this is possible with pure asterisk on a decent system? Is there anything I shoudl watch out for? Your help is much appreciated! Chris
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