<div dir="ltr">Probably best to send the call into freeswitch2 and route it to the user within the dialplan.<div><br></div><div>You can also share registration between the servers which would let you call them from freeswitch1 directly. However many firewalls, especially with NAT might not allow you to do so from anywhere except the server they registered to. Therefore routing via freeswitch2 would be more reliable.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On 24 September 2014 08:48, akhil garg <span dir="ltr">&lt;<a href="mailto:akhilgarg7@gmail.com" target="_blank">akhilgarg7@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Hi,</div><div><br></div><div><br></div><div>                                                                                ---------------------                                                                        --------------------</div><div>                                                                                | Freeswitch 1 |                                                                         |Freeswitch 2 |</div><div>                              ------------------------------------&gt;     |                      |-----------------------------------------------------------&gt;|                      |&lt;------------------------------------------</div><div>                          (user 100 registered to FS1)         |                      |                                                                         |                      | (user 200 registered to FS2)</div><div>                                                                                --------------------                                                                         --------------------</div><div><br></div><div>Freeswitch 1:</div><div>------------------</div><div>Created a gateway ( with User 2222) in external sip_profile which registers to Freeswitch2</div><div>User 100 registers to Freeswitch1 in external sip_profile.</div><div>Public dialplan is as below</div><div><br></div><div><div>  &lt;extension name=gw_To_fs2&quot;&gt;</div><div>    &lt;condition field=&quot;destination_number&quot; expression=&quot;^200$&quot;&gt;</div><div>      &lt;action application=&quot;set&quot; data=&quot;dialed_extension=$1&quot;/&gt;<br></div><div>      &lt;action application=&quot;set&quot; data=&quot;dialed_domain=IP of FS2&quot;/&gt;</div><div>      &lt;action application=&quot;bridge&quot; data=&quot;sofia/gateway/sip_provider_1/${dialed_extension}&quot;/&gt;</div><div>    &lt;/condition&gt;</div><div>  &lt;/extension&gt;</div></div><div><br></div><div><br></div><div>Freeswitch 2:</div><div>------------------</div><div>User 200 registers to Freeswitch2 in external sip_profile.<br></div><div><br></div><div>Public dialplan is as below</div><div><br></div><div><div>  &lt;extension name=&quot;client_1&quot;&gt;</div><div>    &lt;condition field=&quot;destination_number&quot; expression=&quot;^200$&quot;&gt;</div><div>      &lt;action application=&quot;set&quot; data=&quot;dialed_domain=IP of FS2&quot;/&gt;</div><div>      &lt;action application=&quot;set&quot; data=&quot;dialed_extension=$1&quot;/&gt;</div><div>      &lt;action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;</div><div>      &lt;action application=&quot;bridge&quot; data=&quot;user/${dialed_extension}@${dialed_domain}&quot;/&gt;<br></div><div>    &lt;/condition&gt;</div><div>  &lt;/extension&gt;</div></div><div><br></div><div><br></div><div>Query:</div><div>---------</div><div>I am able to call 200 from user 100 using gateway (configured on Freeswitch1 and registered on Freeswitch2)</div><div><br></div><div>How can I call User 100 from from user 200 without confuring a gateway on Freeswitch2? </div><span class="HOEnZb"><font color="#888888"><div><br></div>-- <br>regards,<br>akhil
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