<div dir="ltr"><div><div><div><div><div>Hi,<br><br></div>I plan to use Freeeswitch as Media server for Kamailio and in my case Freeswitch should receive call and loopback it back to Kamailio.<br><br></div>In my case I needed to make do not change: to, from and request-uri headers. I achieved it by:<br><br><action application="bridge" data="[sip_from_uri=${sip_from_uri},sip_invite_to_uri=sip:${sip_to_uri}]sofia/internal/${sip_req_uri};transport=tcp;fs_path=sip:kamailioserver"/><br><br></div>Thank you!<br><br></div>With kind regards,<br><br></div>Jurijs<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-08-14 11:39 GMT+03:00 Stanislav Sinyagin <span dir="ltr"><<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">there's Kamailio and OpenSIPS for that, why do you need that on FreeSWITCH?<br>
<br>
On Wed, Aug 12, 2015 at 11:06 AM, Jurijs Ivolga <<a href="mailto:jurij.ivo@gmail.com">jurij.ivo@gmail.com</a>> wrote:<br>
> Hi,<br>
><br>
> I need to receive call on Freeswitch and Loopback it to original server from<br>
> where it came into, without changing any SIP header(to, from, SIP request<br>
> and etc). Is there easy way to do this?<br>
><br>
> Thank you!<br>
><br>
> With kind regards,<br>
><br>
> Jurijs<br>
><br>
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</blockquote></div><br></div>