[asterisk-users] Completing my Configuration

Tue Sep 25 04:19:29 CDT 2007

Dear Anselm,

I am sorry about the "big traffic" in the newsgroup.

I tried to send my post to the newsgroup  for 3 days now - once a day,
but it did not appear. Today I tried putting it in "cc" also with, then it worked out ...

I will carefully read your answer.

thank you veryy much

-------- Original-Nachricht --------
> Datum: Tue, 25 Sep 2007 09:34:14 +0200
> Von: Anselm Martin Hoffmeister <anselm at hoffmeister-online.de>
> An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Betreff: Re: [asterisk-users] Completing my Configuration

> Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
> > Hallo Group,
> > 
> > I have basically set up a small asterisk system,
> > which ahs 4 peers:
> > 
> > * registers at 2 Sipgates
> > * 2 hardware phones connected to it
> > 
> > Both Hardware phones can phone outwards(cheaper sipgate is selected with
> dialplan)
> > Calls from both sipgates make my hardware phones ring
> > 
> > But here comes the challenges:
> > 
> > Is it possible to configure asterisk in such a way that in the phone:
> > 
> > * there are names instead of numbers in my hardware phone displayed
> 
> Depends on the hardware phones. In theory, with each SIP call connecting
> to the phone, both a name and a number can be transferred. AFAIK sipgate
> defaults to setting both to the usual callerID. That is exactly the
> reason why you can set the variables ${CALLERID(num)} and
> ${CALLERID(name)}.
> 
> Some hardware phones (I assume, the better ones ;-) display both; my
> Allnet for example seems to only display the name, but store the number
> for the "call back" list. My Fritz!Boxen seem to forward both name and
> number to ISDN devices on the internal S0-bus, just not many ISDN phones
> can actually display text "numbers".
> 
> Let your asterisk have an ast database, looking like
> callerid/420123456789 => "Doe, John Q."
> callerid/492240224922 => "Mustermann, Dr. Peter"
> 
> Then you could expand your dialplan logic a little. If you have a line
> 
> exten => 12345,4,Dial(SIP/phone1,60)
> 
> or whatever that looks like in your SIP-incoming context, insert those
> lines before it [and change the "4", "5", "6", "7"s ;-) ]
> 
> exten => 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})})
> exten => 12345,5,GotoIf($["${CALLERID(name)}" = ""]?6:7)
> exten => 12345,6,Set(CALLERID(name)="-- ${CALLERID(num)}")
> exten => 12345,7,Dial(SIP/phone1,60)
> 
> Line 6 treats the case that the number is not in your database and sets
> the callerid-name to "-- NUMBER_OF_CALLER"
> 
> You can manually add data to the astdb from the asterisk CLI with
> 
> database set callerid 420456789 "Silly, Roger M."
> 
> You should check that both your SIP providers provide incoming CLI in
> the international formatting, without country prefix or "+". In my
> experience some SIP providers send numbers like
> 492240224922, others send +49... or 0049..., some send national format
> 02240... for all national calls, some even omit the leading "0" there,
> and some just change the behaviour depending from which network (T-Com
> landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign
> callers...) the call originates. If you have more than two providers,
> this can be a PITA - you will need some dialplan logic to sanitize the
> callerid in those cases, and sometimes you are just left for guessing,
> for example when the provider signals calls from T-Mobile as 16177554224
> and calls from Boston, MA, USA the very same. Germany does not have
> fixed-length numbers, even in the mobile phone networks the length
> differs, and the number given might be valid for both circumstances.
> </rant>
> 
> > * The Ringtone is different for special call numbers 
> 
> If your phone supports that, yes, you can do it. The common method for
> this seems to be sending an additional header. There will be docs on
> "SIPAddHeader(blah)" or similar on www.voip-info.org, and you might want
> to also use a database here to find out wether special ringtones are to
> be activated or not.
> 
> > * it is displayed, in which sipgate the call came from
> 
> You could use the CALLERID(name) field for that, by adding the provider
> short name in front of the caller's name, like
> 
> exten => 12345,4,Set(CALLERID(name)=at-${DB(callerid/${CALLERID(num)})})
> 
> for calls via the "at" provider - or whatever seems stylish enough.
> 
> I personally have a logic that makes use of the dial-around prefix in
> use here in Germany: From a regular T-Com landline you can select the
> provider that will carry the next call by dialling 010[1-9]X or 0100XX.
> Those prefixes of course do not work on SIP provider lines, and my
> asterisk does not have landlines connected. So I use those for my own
> purposes, e.g. selecting the SIP account that the call may go out
> through. Dialplan logic detects "010XX" (100 possible accounts are
> enough, I just ignore 0100XX as additional number field here) and
> selects the outgoing provider accordingly.
> 
> If I wished to have the incoming line signalled to me, I would prefix
> the incoming CALLERID(num) with the provider code. Callbacks would go
> through the same line - nice bonus. Most of my phones do not handle text
> and number simultaneous display in a reasonable way, so I do not rely on
> the text.
> 
> > * using an extension in my call number redirects the call just to one
> >   sip phone ?
> 
> AFAIK you could only do this by Answer()ing the line (at which point the
> caller starts paying the connection) and asking the caller to input an
> extension. (Hint: "Read()"). I personally do not like this solution at
> all, because that is what DID and number block allocation were invented
> for. You can get a number block with SIP from some providers. Or you
> just get yourself another "private" phone number ;-)
> 
> BR,
> 
> Anselm
> 
> 
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