asterisk users - Mar 2021 - CallerID presentation - presentation prohibited but still passing number

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous <anonymous>" as opposed to letting the
receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)




-----Original Message-----
From: phreak at phreaknet.org <phreak at phreaknet.org>
Sent: March 11, 2021 15:33
To: Mike <michael at virtutel.ca>; asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] CallerID presentation - presentation
prohibited but still passing number

I've been able to pass presentation status between tandems without needing
to do anything explicitly. This seems to be part of the Caller ID that is
transmitted without explicit intervention. Have you tested without using
the u option? I've never used the u option and not had issues with
presentation transmitting as supposed to. This is with manual
Set(CALLERID(pres)=something) and then seeing if it gets honored on the
other end. The remote Asterisk gets the full number, of course, but the
called telephone does not display it. Perhaps the u option is intended for
lines, not trunks, and so the number never gets sent?

The only time I've found I (may) need to explicitly account for
presentation is if I am regenerating the call, and then this needs to be
accounted for in the call file (ignore that if that makes no sense).

The only inconsistency I've encountered has to do with presentation
mismatches between the name and the number. If, for instance, I want the
number to display but not the same, setting the presentations as the
documentation would suggest does not work. The behavior is inconsistent
between different SIP clients and it didn't work for me in any logical
way. I didn't bother to a file a bug report about it, as I worked around
this by simply doing Set(CALLERID(name)=) to empty the name and write the
original name back into the variable after the call. Your mileage may
vary.

NA

On 3/11/2021 2:22 PM, Mike wrote:
>
> Hi,
>
> Using Asterisk 13.36.0
>
> I have a bit of a technical issue with hidden caller IDs.  My setup,
> at the moment, is composed of two Asterisk boxes. In some instance,
> calls arrive on Asterisk A, and are then sent to Asterisk B for
> further processing. The link between them is SIP (both on the same
> switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI
link.
>
> When I receive a hidden number (i.e. “presentation prohibited”) call
> on Asterisk A through PRI, I get the following Caller ID information
> (using 444-555-6666 as example):
>
> “ <4445556666>”
>
> And
>
> CallerID presence is received as “prohibited_not_screened”.
>
> Which is fine – I know the incoming number BUT I am told not to show
> it to the end user. All good.
>
> The problem is when calls are not processed on Asterisk A, but sent to
> Asterisk B for further processing. The dial command I used on Asterisk
> A to send calls to AsterisB is the following:
>
> exten => s,n,Dial(SIP/AsteriskB/123,,f(""
> <4445556666>)u(prohib_not_screened))
>
> Again, so far so good. But, on Asterisk B in the appropriate context,
> on extension 123, my first command is a Verbose to show Callerid(all)
> and the received called id is shown as “Anonymous <anonymous>” with
> CALLERID presence still “prohib_not_screened”. I would like Asterisk B
> to receive the actual callerid (“ <4445556666>”) along with the
> appropriate CallerID presence value (which is correct already).
>
> Basically I want to “pass forward” both CALLERID and CALLERIDPRES
> exactly as received on AteriskA to AsteriskB so that AsteriskB gets
> the exact same info AsteriskA had in the first place.
>
> How do I accomplish this?
>
> Michael
>

On Thu, Mar 11, 2021 at 4:50 PM Mike <michael at virtutel.ca> wrote:
> Thank you for taking the time.  I believe you misunderstood my question.
> Callerid presence is passed perfectly already, as shown through Verbose
> commands on both sides of the SIP call. The CALLERID name and numbers
> aren't passed properly ONLY when presence is "hidden".
>
> As if Asterisk decided that since this is a hidden number, to replace the
> number with "Anynomous <anonymous>" as opposed to letting
the receiving
> Asterisk process it as desired with whatever logic I choose.
>
> I just tested without any u() or f() or s() functions - same result. No
> improvement or degradation with my issue. (Not sure why I had these
> options)
>
You probably want to set the "trust_id_outbound" option[1] to
"yes".

[1]
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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On 3/11/2021 2:50 PM, Mike wrote:
> Thank you for taking the time.  I believe you misunderstood my question.
> Callerid presence is passed perfectly already, as shown through Verbose
> commands on both sides of the SIP call.
I should have mentioned before that the scenario I mentioned was with 
IAX2 trunking. I don't use SIP for inter-switch trunking but I do for 
terminating lines. Maybe IAX2 and SIP handle Caller ID
differently.
> The CALLERID name and numbers
> aren't passed properly ONLY when presence is "hidden".
Sorry, by this you mean that Asterisk does not see the number or that 
the telephone set does see it? It sounded like the former.
One thing to note is that prohib and unavailable function differently.
I looked at my code, and if a subscriber does not subscriber to Caller 
ID, I do something like the following:
same => 
n,ExecIf($["${GOSUB_RETVAL}"="1"|"${GOSUB_RETVAL}"="2"]?NoOp():Set(CALLERID(pres)=unavailable))

unavailable and prohib options are interpreted differently by different 
SIP clients (e.g. ATAs or softphones). I did some testing and found that 
there was inconsistent behavior with one option vs. the
other.
> As if Asterisk decided that since this is a hidden number, to replace the
> number with "Anynomous <anonymous>" as opposed to letting
the receiving
> Asterisk process it as desired with whatever logic I choose.
It may not be Asterisk but the telephone itself. I tested with a soft 
phone and it shows Anonymous <anonymous>. However, an actual telephone 
set shows "PRIVATE", regardless of which option I choose. But, this is
just a hunch, and I'm not certain.
> I just tested without any u() or f() or s() functions - same result. No
> improvement or degradation with my issue. (Not sure why I had these
> options)