See brian's example on the list for something that might handle your situation.<br><br>I'm not trying to explain away the need for what you want I just want to make sure you understand<br>how we have things designed.<br>
<br>You may want to join our irc channel for more realtime interaction.<br>#freeswitch on <a href="http://irc.freenode.net">irc.freenode.net</a><br><br><br><div class="gmail_quote">On Sun, Apr 6, 2008 at 5:23 PM, kokoska rokoska <<a href="mailto:kokoska.rokoska@post.cz">kokoska.rokoska@post.cz</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Thank you, Anthony, for your answer. I will write my comments in the<br>
e-mail body.<br>
<br>
Anthony Minessale napsal(a):<br>
<div class="Ih2E3d">> The transfer app sends the same call to some other extension at some<br>
> other dialplan within the switch.<br>
> We are not a sip specific switch so we have to abstract many concepts.<br>
><br>
<br>
</div>Yes, I know it. But I just ask :-)<br>
BTW: Very similar messaging concept is IMO used not only in SIP but in<br>
ISUP (and thus in real telephony world) too.<br>
<div class="Ih2E3d"><br>
<br>
> If you are trying to redirect the call we have 2 ways<br>
> the redirect application<br>
><br>
> <action application="redirect" data="sip:user@host"/><br>
> This will result in a 302 redirect in the case of a sip channel<br>
><br>
<br>
</div>I saw this app. But I want to forward the call, not redirect.<br>
<div class="Ih2E3d"><br>
<br>
> <action application="deflect" data="sip:user@host"/><br>
> This will result in a REFER to the specified uri<br>
><br>
<br>
</div>REFER is IMO used for replacing of one leg of call with another one<br>
(RFC3515), i.e. for "transfer".<br>
<div class="Ih2E3d"><br>
<br>
> That is the only 2 ways we support, if you are looking to divert like a<br>
> proxy does<br>
<br>
</div>Yes, this is what I have to do, because I need to be "transparent" for<br>
calls come from PSTN and than go back to PSTN too (ISUP->SIP->ISUP).<br>
<div class="Ih2E3d"><br>
> remember we are only a b2bua and do not perform proxy features,<br>
<br>
</div>OK. This is big advantage for me, because I need B2BUA :-)<br>
But what I want is possible with B2BUA. Even with Asterisk, but only<br>
with very ugly hacks I hope I am avoid with Freeswitch.<br>
<br>
> see<br>
> openser for that.<br>
><br>
<br>
I'm intensive user of OpenSER. In my current setup it is used as<br>
loadbalancer, registrar, presence server etc. But I realy need B2BUA and<br>
ability to manipulate with media (with much more sophisticated way than<br>
SEMS does).<br>
<br>
<br>
Thanks again for your replay, regards<br>
<div><div></div><div class="Wj3C7c"><br>
kokoska.rokoska<br>
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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