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 <[EMAIL PROTECTED]> > An: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@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 > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users