So you are sure it has NOTHING to do with extensions.conf. This clearly shows your absolute ignorance about what poster is asking and how FreePBX works. Had the problem code been posted, this problem would already have been solved by now.<div>
<br></div><div>And sorry if you think this is policing. You can think whatever you like.</div><div><br></div><div>--</div><div>Zeeshan<br><br><div class="gmail_quote">On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere <span dir="ltr"><<a href="mailto:jeff@sunfone.com">jeff@sunfone.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="im"><br>
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:<br>
> This is not elastix or FreePBX forum and asking non-asterisk related<br>
> questions here is misusing this mailing list. Allow anonymous sip is<br>
> not an asterisk feature. Look in the code in extensions.conf what it<br>
> is programmed to do and you'll figure out why it is happening. Or<br>
> maybe post the code and ask why such a behaviour, which'll be better<br>
> way to ask this elastix related question here. If you know what this<br>
> part of dialplan does, rest is easy to figure out.<br>
><br>
><br>
> Zeeshan A Zakaria<br>
><br>
<br>
</div>Heh - listen to you - top posting, bad english, and self appointed list<br>
police. His problem certainly seemed asterisk related to me, and has<br>
NOTHING to do with code in extensions.conf. He even posted CLI commands<br>
he is attempting to use to find his problem. I applaud him for taking<br>
the initiative to try working it out on his own, and see no problem at<br>
all with his question. I hope we can help him fix it.<br>
<div class="im"><br>
j<br>
<br>
> --<br>
> <a href="http://www.ilovetovoip.com" target="_blank">www.ilovetovoip.com</a><br>
><br>
> > On 2010-09-10 11:17 PM, "bruce bruce" <<a href="mailto:bruceb444@gmail.com">bruceb444@gmail.com</a>> wrote:<br>
> ><br>
> > Hi Everyone,<br>
> ><br>
> ><br>
> > I have a provider whose DID used to come into the box just fine but<br>
> > recently stopped working. Nothing has been changed on our end.<br>
> ><br>
> ><br>
> > Here is what I get when doing "sip set debug peer PROVIDER":<br>
> ><br>
> ><br>
> > Sending to 123.123.123.123 : 5060 (no NAT)<br>
> ><br>
> ><br>
> > ^^^^ That is ALL I am getting with sip debug turned on.<br>
> ><br>
> ><br>
> > With Allow Anonymous SIP set to YES, then the call comes in properly<br>
> > and you see the ACK, REQUEST and ACCEPT of sip debug just fine.<br>
> ><br>
> ><br>
> > This is Elastix with Asterisk 1.4.33.1<br>
> ><br>
> ><br>
> > Any thoughts?<br>
> ><br>
> ><br>
> > Thanks<br>
> ><br>
> ><br>
> ><br>
> > --<br>
> > _____________________________________________________________________<br>
> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a><br>
> > --<br>
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><br>
</div>> --<br>
<div class="im">> _____________________________________________________________________<br>
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><br>
> asterisk-users mailing list<br>
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<br>
<br>
<br>
</div>--<br>
<div><div></div><div class="h5">_____________________________________________________________________<br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Zeeshan A Zakaria<br>
</div>