<div class="gmail_quote">On Thu, May 5, 2011 at 10:53 PM, Duong Manh Truong <span dir="ltr"><<a href="mailto:ngoahotanglongbk@gmail.com">ngoahotanglongbk@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi all, <div>I've created sip trunk on Asterisk and defined asterisk server ip on address table of opensips</div><div><br></div><div>Then, from extension of Opensips , i can dial out to pstn through Asterisk</div><div>
<br></div></blockquote><div><br></div><div>Remember, just because you can dial out, doesn't mean that you are properly registered. Also, depending on your configuration, there are many ways you could be routing this call. I assume you are expecting the call to be routed to the registered device. Have you checked it's registration status with:</div>
<div>opensipsctl ul show</div><div><br></div><div>If it is registered, then the problem is in your configuration and we wouldn't really be able to help you out much without seeing what exactly you've got setup. </div>
<div>-Brett</div><div><br></div></div>