2009/3/24 Christian Victor <[email protected]> > 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 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
If the switch is fine why not ? But i wander why kind if switch is that 240-480 fxo ? ;) Sounds like a big overkill. And i dont see a problem with asterisk, if not too much transcoding involved and with the right hardware.
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