<br><br><div class="gmail_quote">2009/3/24 Christian Victor <span dir="ltr"><<a href="mailto:christian@victormedia.de">christian@victormedia.de</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi!<br><br>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.<br>
<br>Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx".<br><br>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?<br>
<br>Your help is much appreciated!<br><br>Chris<br>
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<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br>If the switch is fine why not ? But i wander why kind if switch is that 240-480 fxo ? ;)<br>
Sounds like a big overkill.<br>And i dont see a problem with asterisk, if not too much transcoding involved and with the right hardware.<br>