<p>Actually it is a very easy to understand and fix issue, but looking at the code taking care of anonymous sip calls is the key. Those who post third party GUI related issues should at least post the underlying asterisk config or code here, so the asterisk part of the problem can be fixed.<br>
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<p>Zeeshan A Zakaria</p>
<p>--<br>
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<p><blockquote type="cite">On 2010-09-11 7:22 PM, "Paul Belanger" <<a href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a>> wrote:<br><br><p><font color="#500050">On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere <<a href="mailto:jeff@sunfone.com">jeff@sunfone.com</a>> wrote:<br>
>> Sending to 123.123.12...</font></p><p><font color="#500050">> Either you changed the peer parameters or they did...<br>></font></p>If he is not receiving any response, it is most likely a routing issue.<br>
<font color="#888888"><br>
--<br>
</font><p><font color="#500050">Paul Belanger | dCAP<br>Polybeacon | Consultant<br>Jabber: <a href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a> | IRC: pabelanger ...</font></p><p><font color="#500050">-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
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