<div>looks like something wrong with the dial plan in the extensions.conf.. i would recommend start debug on and see the content of "full" log may be that give some clue.</div>
<div> </div>
<div>Thanks,</div>
<div> </div>
<div>Vivek<br><br> </div>
<div><span class="gmail_quote">On 11/30/07, <b class="gmail_sendername">Russell Brown</b> <<a href="mailto:russell@lls.lls.com">russell@lls.lls.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>I have two Asterisk systems that can route to each other via a VPN with<br>firewalls disabled for testing purposes.
<br><br>Each Server can see (tested via nmap) UDP port 5060 on the other.<br><br>So... I thought that I could simply use a Dial command in Server A's<br>config to place a SIP call to Server B... but it doesn't seem to work.
<br><br>Server A (<a href="http://192.168.1.33">192.168.1.33</a>) has:<br><br> exten => *136,1,Dial(<a href="mailto:SIP/90@10.10.111.13">SIP/90@10.10.111.13</a>,30)<br><br>but whenever a user on Server A dials '*136' the call doesn't complete
<br>and the CLI shows:<br><br> Executing [*136@from-sip:1] Dial("SIP/112-0071f650", "SIP/90@10.10.111.13|30") in new stack<br> -- Called <a href="mailto:90@10.10.111.13">90@10.10.111.13</a>
<br> -- SIP/10.10.111.13-00793520 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br><br>I can't see anything in Server B's logs from <a href="http://192.168.1.33">192.168.1.33
</a><br><br>What am I missing?<br><br>Any pointers to help me get this working?<br><br>--<br>Regards,<br> Russell<br>--------------------------------------------------------------------<br>| Russell Brown | MAIL:
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