Hi, all<br>i have used opensips as registrar.<br>my scenario,<br><br>opensips->asterisk(routing logic)->opensips<br><br>i have done with opensips to asterisk call .<br>asterisk deside where to call go , and if local call then go to opensips.<br clear="all">
<br>asterisk to opensips call not done.<br><br>any suggetion?<br><br>-- <br>Bhrugu Mehta<br>Sr. S/W Engineer (D&D)<br>VOIP,Telephony Team<br>India<br>