asterisk users - Sep 2010 - 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?

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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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

--
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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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. 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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On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce <bruceb444 at 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.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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 and everyone
clear why is it happening. But poster apparently not well verse in Asterisk
anyways. FreePBX has their own forum as well.

Or maybe you can explore FreePBX code for him.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-11 9:40 AM, "Paul Belanger" <paul.belanger at
polybeacon.com> wrote:

On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce <bruceb444 at gmail.com>
wrote:
> I have a provider whose...
Have you considered contacting your provider?  I would think that is
your first step.

--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

--

_____________________________________________________________________
-- Bandwidth and Colocation Pr...
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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 123.123.123.123 : 5060 (no NAT)
> 
> 
> ^^^^ That is ALL I am getting with sip debug turned on.
> 
I think this may be because the peer is not be recognized as a peer.  If
you know the IP of the source of the call (the provider) try "sip set
debug ip X.X.X.X".  Then you will probably see the rejection.  Not that
that will help you much :)  You need to find out why it is being
rejected.  Either you changed the peer parameters or they did...

j

On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere <jeff at 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
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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

--
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.

--

Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger ...

-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a...
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On Sep 10, 2010, at 10:07 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
> 
Try adding 'insecure=port,invite' to the sip peer definition. If that
doesn't work, you can try 'insecure=very'.

Otherwise, try getting a packet capture using tcpdump on the interface that you
world normally connect to your provider with in order to see all of the sip
traffic that you're missing with just a peer debug from within asterisk.
Once you have that, if the answer doesn't jump out at you, you can post it
here and someone should be able to help.

Thanks,
--Warren Selby