<div dir="ltr">I will let others fill in the details but I think this is what you want to achieve on a high level.<div><br></div><div>A user calls the Boss's number which is actually routed to the assistant's phone.</div><div>Assistant answers call and transfers the call to the Boss's "real" extension <a href="https://en.wikipedia.org/wiki/Call_transfer">https://en.wikipedia.org/wiki/Call_transfer</a>.</div><div><br></div><div>Does that sound like what you want?</div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Nov 2, 2015 at 7:14 AM, Michele Pinassi <span dir="ltr"><<a href="mailto:michele.pinassi@unisi.it" target="_blank">michele.pinassi@unisi.it</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
in my context, i have OpenSIPS as a SIP router and Asterisk as a media<br>
box for IVR, Voicemail etc etc etc...<br>
<br>
I need to implement Secretary/Boss function, like this:<br>
<br>
(USER) ----call--///--> (Boss phone) --- call diverted to secretary---><br>
(Secretary) ----- divert ----> (Boss)<br>
<br>
User cannot call directly to Boss phone: firstly, the call will be<br>
diverted to secretary and only in a second time, call should be<br>
forwarded to boss phone (from secretary).<br>
<br>
Any hint how to do this ?<br>
<span class="HOEnZb"><font color="#888888"><br>
Michele<br>
<br>
--<br>
Michele Pinassi<br>
Responsabile Telefonia di Ateneo<br>
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena<br>
tel: 0577.(23)5000 - <a href="mailto:centralino@unisi.it">centralino@unisi.it</a><br>
<br>
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, <a href="http://www.faq.unisi.it" rel="noreferrer" target="_blank">http://www.faq.unisi.it</a><br>
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<br>
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<br></blockquote></div><br></div>