[Asterisk-Users] Little confused about Caller ID

C F shmaltz at gmail.com
Sun Jan 9 18:16:42 MST 2005 

He has a similar story:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082034.html



On Sun, 9 Jan 2005 20:14:05 -0500, C F <shmaltz at gmail.com> wrote:
> 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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