Hi Yogesh,

Here what we want to do with the switch. Our scenario is like few wholesale carrier will sendus traffic for different destinations, which we will be sending to thedifferent vendors. So we were using 206.222.8.58 as our live switch to testfreeswitch. So we need to understand howfreeswitch is terming following things and where do it makes them so we cankeep adding more customers and vendors. Once it starts working for us. Scenario 1 customer name “Allianz” will be sending call to us from IP 209.62.77.114. presently we can configure this customer as a SIP but we also want to do h323 there. (we don’t authenticate any customer for user/pass, they only come with IP and prefix) they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them. for India(91) calls should go to indiavendor1 on ip address 122.252.236.10 as a 1st preference and once that is fill it sld go to indiavendor2 on IP address 122.252.236.11 calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1.

This bit's straightfoward enough. You can either do this using the dialplan, which means that you'll need to update it when your routing changes, or by calling a script (Javascript) or an external application using the socket interface to do the call routing. We run with a single SIP profile for inbound calls, and then sort out who they're from and what to do with them in an external application using the socket interface, which works fine for us.

Scenario 2 one SIP user created on freeswiotch which we can use to make test calls to the Gateways(vendors) we add to the switch for the testing purpose. that SIP user should be able to call any existing vendors/gateways for different destinations and also new one we will add in future so I think this one will have different dialplan.

We do this by having the user authenticated by IP, and sending them in the same direction as any other inbound call. The application above allows routes to be specified using prefixes, so the test call requirement just falls out of it.

Scenario 3

1. where do we have to create SIP users which we can give it to retail customersfor them to cone in any ip, means they have to have user/pass and ipprovided by us to let them register in our system and once done they can make acall out

This we do dynamically using mod_xml_curl; you could rebuild a directory file each time you add/remove/change a customer and have FS reload it, but - depending on its size and the frequency of changes - that might get a bit unwieldy. The way we do things is by no means the only one; it's probably not even the best one. But it works well, and I can understand it..! Cheers -- Dave

NOTE: where it will dump CDR for all above calls ? Thanks Yogesh.

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