Hi we have a PBX that manage 50 users. and now we want to implement a * server so we can connect to our parthner using SIP in usa to place call. we should have 8 simultanious call. we want to know witch interface we need connect the * server to the PBX so users can place call directly from their phone wich are connect to the PBX.
regards.
OK just to make sure I understand correctly, you have a traditional PBX system with about 50 users. Your business partner is in the USA and you want to be able to call him via SIP.
(curious- are you sending the SIP call straight to him or does it go thru an ITSP, so you are just using SIP to lower your phone bill?)
First you need to connect * to your PBX. How this is done depends on the pbx, namely what kind of interfaces and free ports it has, as well as what kind of programming you can do to it.
The most efficient (and most expensive) way would be to add a PRI interface to the * box and use that. PRI will give you up to about 23 channels (simultaneous calls). Set the PRI as an outbound trunk for the PBX, and program the PBX to route calls to *. Setup * accordingly.
The other way to do it would be to get Asterisk a handful of FXS interfaces. Digium’s TDM400 or Sangoma A200 products can both do this. You need 8 ports. Connect these to spare FXO (trunk/CO/line) plugs on the PBX, and group them together. Program the PBX to use this group for calls to your american partner.
The last way to do it should be avoided if possible. If you don’t have 8 FXO ports on the PBX and have no T1, you can use extension ports. Group 8 of them together and connect them to FXO interfaces in the * box (again, need TDM400 or A200 cards). Program an extension on the PBX that will dial this group. Asterisk should immediately answer an incoming call on any port and send it to DISA(), which will provide a dialtone. User will dial extension (or push button) to get a line on this group, will hear a dialtone, and then will dial his destination.
Hope that helps!