This might help to answer poster's question. It tells how the allow
anonymous sip connections work in FreePBX, and shows the code.
http://www.geekzone.co.nz/sbiddle/7183
http://www.geekzone.co.nz/sbiddle/7183--
Zeeshan
On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger
Thanks guys. I wasn't able to collect enough SIP debug as the problem was
resolved as I was testing different configuration for the trunk. Probably a
change on the provider side.
John Novack: Unfortunately, it seems that this list has a non-stop list of
people who like to stir up things or try to
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
Allow anonymous SIP and enable debug then check if calls coming from
same IP which you have configured in peer?
Regards,
Faisal Hanif//
On 9/11/2010 8:07 AM, bruce bruce wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working.
Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list.
Since the poster may not be sure this isn't an Asterisk problem, and
Asterisk IS involved, your position is unreasonable.
Self appointed list police
On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce bruceb...@gmail.com wrote:
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.
Have you considered contacting your provider? I would think that is
your first step.
--
Mr. John,
This is not about policing and this is asterisk-user mailing list. Poster is
a FreePBX user. I am very well aware of Asterisk IS involved, but the fact
is this is not a FreePBX mailing list. If the poster examines the problem
code from extensions.conf, or post it here, it'll made him
On Fri, 2010-09-10 at 23:07 -0400, bruce bruce 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
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
I think this may be because ...
So you think, don't know. Maybe you knew if you knew the FreePBX code, or
bothered to look into it.
j
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
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
On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote:
Sending to 123.123.123.123 : 5060 (no NAT)
Either you changed the peer parameters or they did...
If he is not receiving any response, it is most likely a routing issue.
--
Paul Belanger | dCAP
Polybeacon | Consultant
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
--
www.ilovetovoip.com
On 2010-09-11 7:22 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
Sending to 123.123.12...
Either you changed the peer parameters or they did...
If he is
Poster is having problem when he disallows anonymous sip peers. Do you know
at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet
seen the dialplan for FreePBX.
When there is enough detail in the post and I am aware of the problem, I
always try to help. I don't believe in
[snip]
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
On Sep 10, 2010, at 10:07 PM, bruce bruce bruceb...@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
On 09/12/10 07:06, Zeeshan Zakaria wrote:
I think this may be because ...
So you think, don't know. Maybe you knew if you knew the FreePBX
code, or bothered to look into it.
For God's sake, stick a sock in it. Others are attempting to help. You
are not.
--
On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Poster is having problem when he disallows anonymous sip peers. Do you know
at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet
seen the dialplan for FreePBX.
It's very simple to find the
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
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