Adding "fromuser" option in trunk declaration in AST1 has solved all problems though.<br><br>
<div class="gmail_quote">On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky <<a href="mailto:igi-go@ya.ru">igi-go@ya.ru</a>> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div class="Ih2E3d">Rizwan Hisham wrote:<br>> I am having a strange problem. I am using my asterisk server AST1 to<br>> register with another asterisk server AST2 using 2 accounts (2 register<br>> commands in sip.conf). I have made 2 local users in AST1, and configured my<br>
> dialplan in such a way that both local accounts on AST1 use different trunks<br>> to send the call to AST2 server. These 2 different trunks are for 2 accounts<br>> i have registered on AST1.<br></div>> (skiped)<br>
<div class="Ih2E3d">><br>> How can i make asterisk realize it?<br>><br></div>You must enable authentication of INVITE that AST1 send to AST2. Now you<br>have no authentication of incoming INVITE and AST2 make decision about<br>
used account based only on IP address of caller peer.<br><br>Changing insecure=port,invite to insecure=port should help.<br><br>--<br>Best regards,<br>Igor A. Goncharovsky<br><br><br>_______________________________________________<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br>