Why don't you think it is correct
behaviour? The purpose of attended transfer is that you consult with the party before transferring with hooking, otherwise it would be a blind transfer for which there is a blind transfer option.<br><br>Rob<br><br><div><span class="gmail_quote">
On 2/10/06, <b class="gmail_sendername">Moises Silva</b> <<a href="mailto:moises.silva@gmail.com">moises.silva@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
this is a Normal behaviour, nevertheless i dont think is a correct
behaviour. Several weeks ago other user asked the same, i suggested him
to open a feature request on <a href="http://bugs.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">bugs.digium.com</a>, check for that<br>
<br>
regards<div><span class="e" id="q_10951b24a7c42bbf_1"><br><br><div><span class="gmail_quote">On 2/9/06, <b class="gmail_sendername">Thomas Artner</b> <<a href="mailto:artner@gmx.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
artner@gmx.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi!<br><br>I am new with asterisk and I have my first problem with the attended<br>call transfer feature.<br><br>When a call comes in, i take the call and i would like to transfer it.<br>So I press the * button (mapped for the attended transfer in
<br>features.conf) and the number for the receiving extension.<br><br>The receiving extension rings and the call can be taken there.<br>So far so good.<br><br>Now to my problem:<br>If I hook on the handset BEFORE the receiving extension take the call,
<br>the caller from outside will be disconnected and the receiving extension<br>stops ringing.<br>Shouldn't the receiving extension keep on ringing until the call is<br>taken? Independent of hooking on the handset or not!
<br>(as it is with the blind transfer feature)<br><br>The incoming line and all of the extensions are POTS, connected on a<br>tdm400p card.<br><br>I use asterisk 1.2.4 and zaptel 1.2.3<br><br>Hope someone could help me.<br>
<br>Thx,<br>Tom<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com
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