<div>Yes. That maybe true.</div>
<div>Can Asterisk do sip REFER blind transfer.<br> </div>
<div>My configure looks like this </div>
<div> </div>
<div>in sip.conf</div>
<div>[sip_proxy-out]<br>type=peer ; we only want to call out, not be called<br>host=<a href="http://192.168.1.180">192.168.1.180</a></div>
<div>port 5064</div>
<div> </div>
<div>in extension.conf</div>
<div>[default]</div>
<div>exten => 519,1,Transfer(SIP/sip_proxy-out)<br> </div>
<div>It sent back a 302 Move to the other peer with </div>
<div><a href="mailto:sip:sip_proxy-out@192.168.180">sip:sip_proxy-out@192.168.180</a> </div>
<div>as URI.</div>
<div> </div>
<div>There are two out of my expect, one is 320 not REFER. second is the URI. </div>
<div>Can somebody help me out on this. I'll read RFC again in the meantime. </div>
<div>As I remember, B2BUA can do REFER also.</div>
<div> </div>
<div>Thanks!</div>
<div> </div>
<div><span class="gmail_quote">On 6/19/07, <b class="gmail_sendername">BJ Weschke</b> <<a href="mailto:bweschke@gmail.com">bweschke@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">On 6/19/07, Lucian Romi <<a href="mailto:romi.lucian@gmail.com">romi.lucian@gmail.com</a>> wrote:<br>
> In this scenario, how to make asterisk send the invite to<br>> SIP/101@192.168.1.180:5064<br>> instead of<br>> Local/101@default<br>><br>> Thanks<br>><br><br>Asterisk isn't a SIP proxy. As such, you need to use some workarounds
<br>to make what you want to do work. One way might be......<br><br>In extensions.conf :<br><br>[default]<br><br>exten => 101,1,Dial(SIP/101@192.168.1.180:5064)<br>exten => 101,n,Congestion()<br>--<br>Bird's The Word Technologies, Inc.
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