<div dir="auto">Have you read freeswitch documentation?</div><div dir="auto"><br></div><div dir="auto">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.</div><div dir="auto">You need to add an extension on the mentioned file and route to the other asterisks </div><div dir="auto"><br></div><div dir="auto">Super simple example (you can find many on the internet)</div><div dir="auto"><br></div><div dir="auto"><pre class="syntaxhighlighter-pre" style="margin-top:0px;margin-bottom:0px;padding:0px;border:0px;outline:0px;font-size:15px;line-height:1.3;overflow-x:auto;word-break:normal;font-family:monospace;color:rgb(41,41,41)" dir="auto"><extension name="inbound-test">
<condition field="destination_number" expression="^(\+493087710024)$">
<action application="bridge" data="sofia/${domain}/$1@[ASTERISK-4-IP]"/></pre><pre class="syntaxhighlighter-pre" style="margin-top:0px;margin-bottom:0px;padding:0px;border:0px;outline:0px;font-size:15px;line-height:1.3;overflow-x:auto;word-break:normal;font-family:monospace;color:rgb(41,41,41)" dir="auto"> </condition>
</extension> </pre></div><div dir="auto"><br></div><div dir="auto">Something like that.</div><div dir="auto">Read up on extensions and bridge.</div><div dir="auto"><br></div><div dir="auto"><br><div class="gmail_quote" dir="auto"><div dir="ltr" class="gmail_attr">On Tue, 27 Apr 2021 at 01:30, John Tuxies <<a href="mailto:atuxnull@gmail.com">atuxnull@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204)"><div dir="ltr"><div>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</div><div><img src="cid:ii_knndyqql0" alt="ITSP_Scenario.jpg"></div><div><br></div><div>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. <br></div><div>
<div>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).</div><div><div><br></div><div><br></div><div>May i have some help on how to configure this thing please? <br></div></div><div><br></div>
</div><div><br></div><div><br></div></div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>