check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general settings.<br><br><div><span class="gmail_quote">On 7/4/07, <b class="gmail_sendername">satish patel</b> <<a href="mailto:satish_patel_2000_2000@yahoo.com">
satish_patel_2000_2000@yahoo.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;">Dear all<br><br> I have install asterisk
1.2.x and it is working fine my setup is like<br><br>[*]-------[Mediant2k]------------[Avaya]<br><br> Now i want to transfer call in internal extension i have read more document on <a href="http://www.voip-info.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
www.voip-info.com</a> but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task <br><br>regards<br><span class="ad"><br>
<br><p>
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