[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?
Jeff LaCoursiere
jeff at sunfone.com
Sat Sep 11 20:20:49 CDT 2010
> --
> www.ilovetovoip.com
>
> > On 2010-09-11 7:22 PM, "Paul Belanger"
> > <paul.belanger at polybeacon.com> wrote:
> >
> >
> >
> > On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere <jeff at sunfone.com>
> > wrote:
> > >> Sending to 123.123.12...
> >
> > > Either you changed the peer parameters or they did...
> > >
> >
> > If he is not receiving any response, it is most likely a routing
> > issue.
> >
> > --
> >
[un top posting]
On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote:
> 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.
>
>
> Zeeshan A Zakaria
>
>
Its not that he isn't receiving a response - its that his peer debug
statement isn't getting tripped because the peer hasn't authenticated.
That's why I suggested he debug by IP rather than peer. Then what he
will see is the SIP auth attempts and asterisk rejecting them, but in my
experience not much is of value in seeing those packets - it doesn't
point to *why* the connection is being rejected. The routing must be ok
since allowing guest sip connections (the result of setting "accept
anonymous" in FreePBX) allows the calls to come in fine.
His problem is the peer authenticating. This of course has nothing to
do with extensions.conf, as the dialplan is not involved. It is a SIP
authentication problem, purely. There is no "relevant code" to post,
and if you had ever looked into FreePBX's "relevant code" you would
realize that it is actually fairly complex, and you would indeed have a
difficult time debugging the flow.
It *might* help if he posted his peer entry, but without seeing the
other side that may not help much either. As Paul suggested first off,
he should be in touch with his provider, whose tech support should be
able to help him sort it out.
I ran into a strange one EXACTLY like this just last week. We have a
residential dial-tone customer with a Linksys SPA2102 (our standard
device for this service). He had someone come out and replace his home
router, and when he did he stopped authenticating. He has a fixed IP,
so I enabled the debugging as I have mentioned twice now (by IP) and saw
the attempts and rejections. After much hair pulling I *disabled* nat
in his peer entry and it suddenly connected fine. This is bizarre, as
our standard peer configuration works for 100% of the rest of our
customers, who all connect from behind their home nat gateways of all
kinds. I still don't know why that fixed it.
Sorry you took it so harshly Zeeshan, but the only posts that stick out
to me from you are the ones where you are bashing people for posting
questions. I don't recall any off the top of my head where you are
actually helping. Yup, I consider that policing, and it isn't needed.
Like someone else suggested, if you don't want to read it, delete it.
And no, I am not going to bother to read back through archives to see if
that is the truth. Its my impression of your posts, thats all.
j
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