Oh so sorry.<br>Yes you are right, the 'callee'.<br><br>We have one soft switch somewhere located in US.<br>When the call comes, then asterisk has to see the callee number in the sip extensions. If the number is not in its extensions, then that call should be routed to that soft switch in US. But the condition is, a number 4237 should be added in callee's number. For example,<br>
<br>The call comes destined to '123456', the asterisk will see in its sip extensions. If the number is there then the conversation will start right away. But if the extension is not there, then asterisk with route this call to softswitch but 4237 added, that is callee's number will be 4237123456.<br>
<br>I hope you understand me.<br><br>Thanks a lot.<br><br>Regards,<br>Pratik<br><br><div class="gmail_quote">On Mon, Aug 30, 2010 at 10:58 AM, Jose P. Espinal <span dir="ltr"><<a href="mailto:jpe@slackware-es.com">jpe@slackware-es.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hello Pratik,<br>
<br>
Could you please elaborate your question a little more (describe better<br>
your scenario)?<br>
Are you sure to be using properly the terms 'caller', and not instead of<br>
'callee'? (caller : makes call, callee : receives call)<br>
<br>
<br>
Regards,<br>
<div><div></div><div class="h5"><br>
<br>
Pratik Shrestha wrote:<br>
> Dear All,<br>
><br>
> First, I am not so much experienced in Asterisk.<br>
><br>
> I need asterisk to route the call to soft switch when the caller is<br>
> not in its extensions list. And also when routing to soft switch, a<br>
> number 4327 has to be added in the caller's number and then routed. I<br>
> think its not so hard in asterisk. Please help me.<br>
><br>
> Regards,<br>
> Pratik<br>
<br>
</div></div>--<br>
Jose P. Espinal<br>
<a href="http://www.eslackware.com" target="_blank">http://www.eslackware.com</a><br>
IRC: [OFTC|FreeNode]<br>
Khratos @ #slackware | #asterisk/-doc/-bugs<br>
<br>
<br>
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