Have you read freeswitch documentation? All you need to do is create a dialplan in your public.Conf.xml and configure your asterisk to send calls to freeswitch on port 5080. You need to add an extension on the mentioned file and route to the other asterisks Super simple example (you can find many on the internet) <extension name="inbound-test"> <condition field="destination_number" expression="^(\+493087710024)$"> <action application="bridge" data="sofia/${domain}/$1@[ASTERISK-4-IP]"/> </condition> </extension> Something like that. Read up on extensions and bridge. On Tue, 27 Apr 2021 at 01:30, John Tuxies <atuxnull@gmail.com> wrote:
I got an isolated network and i got Freeswitch in it. I took user 1000 and registered it in an Asterisk server. The same happened for accounts 1001-1003 to different servers. Here is an icon of what i am trying to accomplish [image: ITSP_Scenario.jpg]
The network is not connected to Internet or any other Voice service. I would like Freeswitch to act as an ITSP for the Asterisk servers and route in/out numbers. For example Asterisk's_1 server user which has the DID +493087710024 to be able and dial the number +49308771360 and reach another user on Asterisk Server 4. All the communication will go through the Server (Freeswitch).
May i have some help on how to configure this thing please?
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