[asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

2018-06-05 20:29 GMT+02:00 George Joseph <gjoseph at digium.com>:

>
>
> On Tue, Jun 5, 2018 at 10:59 AM Olivier <oza.4h07 at gmail.com> wrote:
>
>>
>>
>> 2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>:
>> Thank  you very much, George for replying.
>>
>>>
>>>
>>> On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> After a long discussion with a friend, I would like to ask here:
>>>>
>>>> 1.According SIP RFCs, is possible/recommended to have different values
>>>> in From and P-Asserted-Id fields ?
>>>> For instance, From field showing 123456789 and P-Asserted-Id showing
>>>> 987654321 (beside privacy considerations) ?
>>>>
>>>
>>> Possible? yes absolutely.
>>>
>>
>> How would you then configure both headers, respectively ?
>>
>> From memory, in previous testings, whenever  CALLERID was set to
>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>> inferred from this that P-Asserted-Id was meant for Privacy
>> considerations and nothing else (see [1])
>>
>
> PAI should be used to indicate the calling party's identification
> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
> function to control privacy on a call by call basis.
>
>
>
I'm sorry but I still have a doubt ...
Let me re-phrase my question:

My setup is:
Asterisk <--- PJSIP ---> Bob

For a reason, I want Bob's phone to receive a call with the following
headers:

From: "Foo" <sip:999 at 1.2.3.4>;tag=as75ee8c7c
P-Asserted-Id: "Foo" <sip:88888 at 1.2.3.4 <sip%3A999 at 1.2.3.4>>;whatever

My dialplan is:
same = n,Set(CALLERID(num)=999)
XXX
same = n,Dial(PJSIP/123456 at bob)

What shall I replace XXX with to allow me to set 88888 in the user part of
P-Asserted-Id URI (see example above) ?
CALLERPRES would change From or P-Asserted-Id but not having different user
parts in URI, would it ?

To my knowledge, a possible way to implement what I'm after is to "turn
off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
PJSIP_HEADER.
Am I missing something ?



>
>
>>
>>
>> [1] https://www.voip-info.org/p-asserted-identity-and-remote-par
>> ty-id-header/
>>
>>
>>> Recommended? who knows?  Implementations are all over the place.  I've
>>> always thought of the From header as identifying the user agent making the
>>> request which kinda agrees with RFC3261.   The PAI header should contain
>>> the identity of the original caller.
>>>
>>>
>>>>
>>>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>>>> Diversion/History-Info header to include Alice's number ?
>>>>
>>>
>>> No.  The diversion header shows who the diverter is.
>>> https://tools.ietf.org/html/rfc5806
>>>
>>

Thank for  this reference: I think I confused diverter/caller/callee roles
when I first read this document.

So, if Bob forwards to Cory a call from Alice, in which headers would you
expect Alice and Bob numbers to respectively appear ?




>
> Best regards
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>>>
>>> --
>>> George Joseph
>>> Digium, Inc. | Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & www.asterisk.org
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
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>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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