<div>Hi </div>
<div> </div>
<div>thanks, would mind pointing to me that</div>
<div>let me check and see</div>
<div> </div>
<div>is that discussion will help me</div>
<div> </div>
<div>ram<br><br> </div>
<div><span class="gmail_quote">On 3/2/06, <b class="gmail_sendername">Paul Hales</b> <<a href="mailto:pdhales@optusnet.com.au">pdhales@optusnet.com.au</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>canreinvite = yes tells the phones to try and talk to each other and<br>leave Asterisk out of the mix.
<br><br>The important word here is TRY.<br><br>There are lots of reasons that it might not quite work, and there was a<br>big discussion on the list about it a little while ago.<br><br>PaulH<br><br>On Thu, 2006-03-02 at 01:55 +0530, ram wrote:
<br>> Hi all<br>><br>> iam working with * just started<br>><br>> can some one explain me canreinvite=yes<br>><br>> when should i use the above options<br>><br>> I would like to use my * server for authentication and directly talk
<br>> SIP user to SIP user<br>> with out consuming my * bandwidth, is that correct<br>><br>> Does any one know, which provider support this option<br>><br>> ram<br>> _______________________________________________
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