[Asterisk-Users] Little confused about Caller ID

Sun Jan 9 18:14:05 MST 2005

Without going into detail of what the query is actualy called, when
the called switch make the query to find out the name, does it ask it
from the originating switch? or it askes it from the switch that is
responsibble for servicing the number?
The difference would be if, I have a PRI from provider A, and a number
from provider B, and thru provider A a make an outgoing call using the
callerID of provider B, who recieves the query?
In my experrience it doesn't matter, b/c most providers will anyhow
ignore the name you send and just send the number (or is it the called
switch that ignores it, and then makes the name query), and since I
don't know how a query like this works I don't know why my provider
doesn't answer with the name I supplied, if it is the one being asked,
I beleive it will not generate names on the fly even if it the one
being queried. But if the calling switch is the one being queried then
it would at least in theory be possible to answer what ever I wanted.
But if it askes the servicing switch (provider B) then it is not even
possible to change it (at least thru A).

I used to work with an Avaya Difinity G3. We had PRI, which gave us
incoming CallerID (Name Only), and outgoing CallerID, since we had DID
we set up the Difinity to send the DID as CallerID, we however also
wanted that the name of the associated extension should come up, we
were then told by our provider that the name will always be the
billing name and there is nothing we can do. One day however a call
came in from another company using a PRI, and a Difinity and the name
of the remote extension came up, which leads me to believe that the
name is sent along on PRI (remember on our PRI we never got incoming
Name on CallerID). But it is ignored by the switches on the way, if
the called end is analog the switch does the query (igonring the name
that is sent, and sending the result of the query as the name), but if
the called end is PRI it does not do a query (I guess it relies on
your switch to do it), however it is sent along, on the PRI end your
switch (in my case the Difinity) just has to read it, since our switch
spoke the same language as the sending switch it received it and the
name showed up on CallerID (we were both using a Difinity).
Some clearfication please.
Thanks.

On Sun, 9 Jan 2005 18:40:33 -0600, Tom Chandler <tchandle at bayou.com> wrote:
> 
> ----- Original Message -----
> From: "Alexander Lopez" <alex.lopez at opsys.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Sunday, January 09, 2005 5:55 PM
> Subject: RE: [Asterisk-Users] Little confused about Caller ID
> 
> > Thanks. I was always under the impression that they were all separate
> > tables in the same DB and that they were collectively called 'The
> > LIDB!!'
> >
> > For my and the others here could you describe the function of the
> > different DBs?
> >
> > I now understand the CNAME, I thought I knew the LIBD, I can guess on
> > the LNP, and 800, but what about the AIN???
> >
> 
> AIN = Advanced Intellignet Network.  This is the area were a large number of
> new
> applications in the SS7 world are going.    Followme calling, and some
> others.
> Some of the auto callback features, speed dial, etc.  These application are
> stored
> in a database, and again require a TCAP query to make them work.
> 
> There are other databases assoicated with Cellular that are completely
> different from the
> TDM world.
> 
> > BTW. I was under the impression the fields had been added to the LIDB to
> > handle the Do Not Call list. Can anyone confirm or deny??
> 
> I have not seen any documentation on this.  I think a large number of people
> get
> LIDB and the Line Record Set which is in the switch interchanged.  The Line
> Record Set
> is the record in the switch which configures the 7 digit line.  There are
> many flags in this record.
> You can block Caller ID, Caller Name, auto callback, block out going LD,
> etc.  The LIBD record
> is in an SCP. The LIBD record and the Line Switch record are two different
> items.
> 
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom
> > Chandler
> > Sent: Sunday, January 09, 2005 6:54 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Little confused about Caller ID
> >
> > TCAP is a transaction application.  The CNAME, LIDB,800,.LNP and AIN
> > database COULD
> > be in the same SCP, but in most cases it is not.  LIDB database are used
> > for
> > calling card, operator
> > services, etc.  These are all seperate databases stored for use in an
> > SCP
> > connected to STP's.
> > So is there a relationship between CNAME and LIBD, no.
> > Tom C.
> >
> > ----- Original Message -----
> > From: "Alexander Lopez" <alex.lopez at opsys.com>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Sunday, January 09, 2005 5:44 PM
> > Subject: RE: [Asterisk-Users] Little confused about Caller ID
> >
> >
> > > Is the TCAP DB part of the LIDB collective (no Borg pun intended)??
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom
> > > Chandler
> > > Sent: Sunday, January 09, 2005 6:45 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion; C F
> > > Subject: Re: [Asterisk-Users] Little confused about Caller ID
> > >
> > > Caller Name is stored in a SCP.  It is a TCAP transaction.  The
> > > receiving
> > > switch via SS7 recieves
> > > the calling party number in the ISUP message of the SS7 datastream.
> > It
> > > is
> > > normally in the IAM mesasge.  Then a TCAP CNAME query is launched from
> > > the
> > > called switch thru
> > > the STP's to a SCP which has the calling name database.  The TCAP
> > query
> > > returns back to the launching
> > > switch the caller name.  LIDB is for operator services etc. CNAME is a
> > > TCAP
> > > database lookup, much
> > > like 800 number translations.
> > >
> > > Tom C.
> > >
> > > ----- Original Message -----
> > > From: "Alexander Lopez" <alex.lopez at opsys.com>
> > > To: "C F" <shmaltz at gmail.com>; "Asterisk Users Mailing List -
> > > Non-Commercial
> > > Discussion" <asterisk-users at lists.digium.com>
> > > Sent: Sunday, January 09, 2005 5:30 PM
> > > Subject: RE: [Asterisk-Users] Little confused about Caller ID
> > >
> > >
> > > > OK here it goes..
> > > >
> > > > Caller ID is two parts or actually three:
> > > >
> > > > Part 1 Number only
> > > > Part 2 Number + Name
> > > > Part 3 Whole lotta stuff (also known as ADSI)
> > > >
> > > >
> > > > Here is the US, I cannot speak for other countries.
> > > >
> > > > When party A places a call to Party B. Party A's Telco picks up the
> > > > number, either from a table on the switch or passed from the PRI
> > from
> > > > Party A.  Then on the far side (Party B's Telco) the Telco does a
> > > lookup
> > > > in the LIDB (Line Information Data Base) and associates a name with
> > a
> > > > number. This information is then passed as Part II CLID.
> > > >
> > > > I have simplified the process, leaving out many processes along the
> > > way
> > > > but it should give some insight as to how the Name actually shows up
> > > on
> > > > the other end.
> > > >
> > > > Most Telcos do not receive the Name as part of the data in the call
> > > > through the tandems b/w Telcos, they opt rather to do the lookup in
> > > the
> > > > LIDB themselves.
> > > >
> > > >
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: asterisk-users-bounces at lists.digium.com
> > > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
> > > > Sent: Sunday, January 09, 2005 6:16 PM
> > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > Subject: Re: [Asterisk-Users] Little confused about Caller ID
> > > >
> > > > When calling to the PSTN (outside VOIP or *) then you will not be
> > able
> > > > to supply the name of callerID even if you have a PRI. The only
> > thing
> > > > you can provide is the number and the receiving switch of the call
> > is
> > > > the one responsibble for attaching a name to the phone number thru
> > > > SS7. If you have a SS7 switch then you could in theory attach the
> > name
> > > > (I have never tried it, but that's what I was told).
> > > > Hope this helps.
> > > >
> > > >
> > > >
> > > > On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette
> > > > <digium at muel.org> wrote:
> > > > > Hi,
> > > > >
> > > > > I've got the Caller ID name and number working with the
> > application
> > > > > SetCIDNumber and SetCIDName.
> > > > >
> > > > > [...]
> > > > > exten => s,3,SetCIDNumber(4183289901)
> > > > > exten => s,4,SetCIDName(Frank Black)
> > > > > exten => s,5,Dial(IAX2/prov01/${DEST})
> > > > > [...]
> > > > >
> > > > > You can also use SetCallerID(Frank Black <4183289901>), but no
> > > success
> > > > for
> > > > > me...
> > > > >
> > > > > bye,
> > > > >
> > > > > Samuel T. Cossette
> > > > > samuel at levinux.org, 1.418.8o2.784o
> > > > > << Well, that's for me to know and you to find out. >> Jeffrey,
> > Blue
> > > > Velvet
> > > > >
> > > > > > Hi Everybody,
> > > > > >     Sure this has been covered a million times on wiki, but
> > > couldn't
> > > > find
> > > > > > an
> > > > > > exact answer to my question.  I am using * to dial out to
> > peoples
> > > > phones
> > > > > > to
> > > > > > give them alerts of different things.  Problem is that the only
> > > > Caller ID
> > > > > > I
> > > > > > can get working is the telephone number.  I am unable to display
> > a
> > > > name
> > > > > > along with the number.  Thinking maybe its the phone receiving
> > the
> > > > call, I
> > > > > > tried my cellphone and my house phone and I can only get the
> > > number
> > > > to
> > > > > > display.  If I leave the number portion out, Caller ID shows
> > > > > > "Unavailable".
> > > > > > Is there a simple way to get a Caller name setup?  I've tried
> > > > examples on
> > > > > > Wiki as well but I couldn't get them to work.
> > > > > >
> > > > > >
> > > > > > ***** extensions.conf *********
> > > > > > [general]
> > > > > > static=yes
> > > > > > writeprotect=no
> > > > > >
> > > > > > [globals]
> > > > > > CONSOLE=Console/dsp ; Console interface for demo
> > > > > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> > > > > >
> > > > > > [sports]
> > > > > > exten => s,1,ResponseTimeout,5
> > > > > > exten => s,2,Answer
> > > > > > exten => s,3,Wait(1)
> > > > > > exten => s,4,Playback(sports/gafanaSports)
> > > > > > exten => s,5,Goto(2000,2)
> > > > > > exten => 2000,1,wait(1)
> > > > > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU)
> > > > > > exten => t,1,Playback(goodbye)
> > > > > > exten => t,2,Hangup
> > > > > >
> > > > > > ******** sip.conf ***********
> > > > > > [general]
> > > > > > context=default
> > > > > > port=5060
> > > > > > srvlookup=yes
> > > > > > allow=ulaw
> > > > > > register => [id]:[pw]@[host]
> > > > > > [gafana]
> > > > > > type=peer
> > > > > > secret=[secret]
> > > > > > username=[username]
> > > > > > host=[hostname]
> > > > > >
> > > > > >
> > > > > > ******* test.call file **********
> > > > > > Channel: SIP/[myNumber]@gafana
> > > > > > CallerID: [My Number]
> > > > > > MaxRetries: 0
> > > > > > RetryTime: 300
> > > > > > WaitTime: 45
> > > > > > Context: sports
> > > > > > Extension: s
> > > > > > Priority: 1
> > > > > >
> > > > > >
> > > > > > What do I need to add to be able to send a name as well and not
> > > just
> > > > a
> > > > > > number?
> > > > > >
> > > > > > Gabe
> > > > > >
> > > > > > _______________________________________________
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