[asterisk-users] Automatic Number Identification and anonymous calls

Fri Feb 10 02:03:45 CST 2012

2012/2/9, Maximilian Grobecker <m.grobecker at portunity.de>:
> Hello,
>
> I know about the german phone system that the sense of an anonymous call
> is, that the called party has no way to get the caller's number in any way.
> The last switch honours the "anonymous" bit and removes the phone
> numbers before sending the call to the called party.
>
> In EURO-ISDN you have a feature called "CLIRO" which means "Calling Line
> Identification Restriction Override" and is only availiable to emergency
> services or special governmental services.
> With this line feature the last switch ignores the anonymous bit and you
> can see both caller id and ANI.
>
> The US phone system may work in another way, but if I call another
> person and want to hide my number I would appreciate if my phone number
> is not sent to the called party in any field.
> So this issue sounds to me more like feature than a bug ;-)

Yes, I agree but what I rate as a bug is "asterisk copying CID data
into ANI field".
The reasoning is asterisk should don't modify, unless explicitely
configured to do so, whatever comes from the public network side.

For instance, it seems analog lines cannot support ANI. So when
receiving a call through an analog line, this ANI field should be
simply empty.
It seems that today default is to copy CID into ANI field. This is
misleading as you may think this ANI is somehow "more trustable" than
CID alone.

I agree that if a caller requires anonymity, then, except for very
special cases (emergency services, for instance), ANI should also be
hidden by the telco.

>
>
>
> Greetings from Wuppertal
> Max Grobecker
>
>
> Am 09.02.2012 11:12, schrieb Olivier:
>> 2012/2/8, Kevin P. Fleming <kpfleming at digium.com>:
>>> On 02/08/2012 12:40 PM, Olivier wrote:
>>>> 2012/2/8, Kevin P. Fleming<kpfleming at digium.com>:
>>>>> On 02/08/2012 10:06 AM, Carlos Alvarez wrote:
>>>>>>
>>>>>> On Wed, Feb 8, 2012 at 2:35 AM, Olivier<oza_4h07 at yahoo.fr
>>>>>> <mailto:oza_4h07 at yahoo.fr>>  wrote:
>>>>>>
>>>>>>      I always thought that ANI (Automatic Number Identification) could
>>>>>> not
>>>>>>      directly be set or changed by end users.
>>>>>>      In this experiment, it seems that if an end user calls
>>>>>> anonymously,
>>>>>>      the ANI is also hidden to the receiving party.
>>>>>>
>>>>>>
>>>>>> I never work with analog lines, but I do believe that anything less
>>>>>> than
>>>>>> a PRI would have the blocking done at the telco end before you get it.
>>>>>
>>>>> Mostly correct; ANI is only delivered over ISDN, SS7 and *some* E&M
>>>>> trunk circuits. Analog circuits have no ability to transport ANI.
>>>>>
>>>>> --
>>>>> Kevin P. Fleming
>>>>> Digium, Inc. | Director of Software Technologies
>>>>> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
>>>>> kpfleming
>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>>>> Check us out at www.digium.com&  www.asterisk.org
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>>
>>>> Thanks for telling ANI feature can't come along analog lines.
>>>>
>>>> I did another test towards an ISDN BRI line and got the same results :
>>>> ANI and CID are either both displayed or both hidden at the same time.
>>>
>>> You need to talk to your provider to see if your circuit is provisioned
>>> to send you ANI at all; if it's not sent by the provider, Asterisk just
>>> duplicates the CLID into the ANI channel variable.
>>
>> Then, may I say that I would prefer to have the ANI left blank, if the
>> network doesn't provide this data, and let system administrators know
>> about this.
>>
>>  Most end-customer
>>> (retail) ISDN circuits are not configured to send ANI unless the
>>> customer asks for it.
>> This matches what I had in mind : ANI being something dedicated to
>> emergency services that may also be available to others requiring it.
>>
>>
>>>
>>> --
>>> Kevin P. Fleming
>>> Digium, Inc. | Director of Software Technologies
>>> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
>>> kpfleming
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>> Check us out at www.digium.com & www.asterisk.org
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
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>>
>
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