[asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

[Zeeshan Zakaria]

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.

And sorry if you think this is policing. You can think whatever you like.

--
Zeeshan

On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere <jeff at sunfone.com> wrote:

>
> On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
> > 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.
> >
> >
> > Zeeshan A Zakaria
> >
>
> Heh - listen to you - top posting, bad english, and self appointed list
> police.  His problem certainly seemed asterisk related to me, and has
> NOTHING to do with code in extensions.conf.  He even posted CLI commands
> he is attempting to use to find his problem.  I applaud him for taking
> the initiative to try working it out on his own, and see no problem at
> all with his question.  I hope we can help him fix it.
>
> j
>
> > --
> > www.ilovetovoip.com
> >
> > > On 2010-09-10 11:17 PM, "bruce bruce" <bruceb444 at gmail.com> wrote:
> > >
> > > Hi Everyone,
> > >
> > >
> > > 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.
> > >
> > >
> > > Here is what I get when doing "sip set debug peer PROVIDER":
> > >
> > >
> > > Sending to 123.123.123.123 : 5060 (no NAT)
> > >
> > >
> > > ^^^^ That is ALL I am getting with sip debug turned on.
> > >
> > >
> > > 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.
> > >
> > >
> > > This is Elastix with Asterisk 1.4.33.1
> > >
> > >
> > > Any thoughts?
> > >
> > >
> > > Thanks
> > >
> > >
> > >
> > > --
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-- 
Zeeshan A Zakaria
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