<p>This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out.<br>
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<p>Zeeshan A Zakaria</p>
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<a href="http://www.ilovetovoip.com">www.ilovetovoip.com</a></p>
<p><blockquote type="cite">On 2010-09-10 11:17 PM, "bruce bruce" <<a href="mailto:bruceb444@gmail.com" target="_blank">bruceb444@gmail.com</a>> wrote:<br><br>Hi Everyone,<div><br></div><div>I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end.</div>
<div><br></div><div>Here is what I get when doing "sip set debug peer PROVIDER":</div>
<div><br></div><div>Sending to 123.123.123.123 : 5060 (no NAT)</div><div><br></div><div>^^^^ That is ALL I am getting with sip debug turned on.</div><div><br></div><div>With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine.</div>
<div><br></div><div>This is Elastix with Asterisk 1.4.33.1</div><div><br></div><div>Any thoughts?</div><div><br></div><div>Thanks</div><div><br></div>
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