[Rtcomm] [Rtcomm] [Bug 6930] Incomming CallerID on VoIP account fails to associate with existing contact

From: bugzilla-daemon at maemo.org bugzilla-daemon at maemo.org
Date: Tue Dec 29 15:59:03 EET 2009
https://bugs.maemo.org/show_bug.cgi?id=6930


mikhail.zabaluev at nokia.com changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
         AssignedTo|rtcomm at maemo.org            |nobody at maemo.org
          Component|Call Application UI         |General
            Product|Chat & Call & SMS           |Contacts
          QAContact|call-ui-bugs at maemo.bugs     |contacts-bugs at maemo.bugs




------- Comment #11 from mikhail.zabaluev at nokia.com  2009-12-29 15:59 GMT+3 -------
(In reply to comment #10)
> Yes, the caller name (or business) is shown on the handset. Of course the
> actual phone number is known too... in case I want to call back. I do not know
> the details how that is made possibe. It "just works" with the Grandstream
> HT-502 SIP adapter and phone that has caller ID.

I see. But this is a bit different problem: not displaying your caller's
display name as reported, along with the SIP URI. You could file this as a
separate bug, but it was already discussed and postponed to Harmattan
internally.

As for the original bug's expectation, an association of a SIP URI to a phone
number in the address book, this is a bit more complex. It can be done in the
address book with some heuristics, and maybe there is not much probability for
a false match as the phone numbers are long enough. But I'm not even sure it's
appropriate: the telephony caller ID, and by association a contact card's
association with a telephone number, has significantly more trust associated
with it than whatever a SIP caller claims itself to be on the internet. Even if
we prominently display the SIP URI, users can easily be misled.
Cryptographically strong identity validation extensions have been published for
SIP, but they seem to be nowhere widely deployed among internet services.

I'm reassigning the bug to Contacts for further considerations of automatic
binding between SIP URIs and telephone number fields.

A somewhat tedious workaround could be, when you receive such a call and you
can map it to an existing contact, to merge the SIP caller's ID with the
contact from the call list view.


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