<div dir="ltr">But bear in mind that the client might not follow the 302. FreeSWITCH for example either terminates the call on 302 or returns to the dialplan so you can validate the new destination, so that you don't (as an example) have someone redirecting you to a premium rate number without the caller's knowledge.<div class="gmail_extra"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On 11 January 2016 at 13:59, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">redirect using full sip uri should do a 302<div><div><span></span><br><br>On Monday, January 11, 2016, ik <<a href="mailto:idokan@gmail.com" target="_blank">idokan@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div>Hello,<br><br></div>I have an incoming call that on some occurrences, instead of answering it, I need to "move" the call to a different Freeswitch server.<br><br></div>I have tried to use deflect, and redirect, however the call with deflect just hangup immediately, and with redirect, it arrive to timeout and then hangup.<br>I'ved noticed at the wiki page (confluence) that the call must be answered with deflect.<br><br></div>Only when I use bridge, the other Freeswitch see's the SIP request and handle the call properly, but as a statful request.<br></div><br></div><br>How can I make the original Freeswitch out of the routing or at least make it stateless redirect (302) instead of stateful one?<br><br></div>Thanks,<br></div>Ido<br></div>
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