Yogesh,

 

These are some serious needs, but I’m pretty sure that FS can be set up to do all of them.  I’m no expert, so I’ll have to defer to the big boys.  However, just from looking at the scenarios I’m guessing that scenario 2 can be done pretty easily in the dialplan and directory.  Just to make sure that we are all on the same page, could you answer a few follow up questions?

#1 – Is your test server up and running right now? 

#2 – Are you running with the default configs?

#3 – Have you made any changes or additions to the default configs?

#4 – What revision are you on?

 

Let’s get those nailed down first and then we’ll take it from there.

 

-MC

 

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Yogesh Patel
Sent: Monday, March 17, 2008 11:05 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] freeswitch Scenario

 

HI

Here what we want to do with the switch.

Our scenario is like few wholesale carrier will send us traffic for different destinations, which we will be sending to the different vendors. So we were using 206.222.8.58 as our live switch to test freeswitch.

 So we need to understand how freeswitch is terming following things and where do it makes them so we can keep adding more customers and vendors. Once it starts working for us.

Scenario 1

  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)
  2. they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them.
  3. 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
  4. calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1.

Scenario 2

  1. 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.
  2.  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.

Scenario 3

 

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

 

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

 Thanks

Yogesh.