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Hi,
first of all thank you for your time and reading this.
This is my first post in this forum and I hope you don’t mind that I post in English. (I’ve been living in Germany for just 5 months but I don’t have difficulties understanding German).
We are a team comprised of 52 attorneys at law based in Frankfurt (in different offices though) and we specialize in advising international clients on various legal issues affecting German law (ranging from criminal and private law, banking to real estate and much more).
Right now we have a lot of clients calling from all over Germany who like to take advantage of our services without traveling to Frankfurt. Hence, I came up with an interesting solution:
Legal counseling via premium rate numbers and VoIP!
I will be setting up German premium rate numbers (0900 and 0180 for prepaid where the fee per minute can be flexibly changed) which our customers can conveniently call. The caller is then introduced to an IVR application, introducing all lawyers (and their specialization) that are logged in to the caller by playing their prerecorded welcome messages. After all welcome messages have been played; the caller is prompted to choose his or her desired lawyer by entering the lawyer’s 3 digit extensions via phone. Alternatively, the caller may also skip the welcome message by directly entering the extension of his or her desired lawyer.
However this application should NOT rely on an ISND/Primary Rate Interface solution (also commonly known as T1 in the US), but rather on a Voice over IP solution powered by Asterisk. The incoming calls from the premium rate numbers would be forwarded via VoIP using the SID protocol, thus allowing us to lower the costs significantly. That’s why I’d like to setup a dedicated Linux server, running Asterisk on it.
I already spoke to our premium rate service number provider (CNS24 AG) and they assured me that they could forward (or route) all incoming calls to my VoiP server (Asterisk) using the SID protocol.
But now to my question: What kind of software (or hardware) do I need to forward the received calls from the Voip Server (Asterisk) to our lawyers which are located in different offices and still use analogue phones (over PSTN)? Do I need a special SID provider or a special Voip Gateway?
Unfortunately I’m not a telecommunication expert, but I’d appreciate if you could tell me the requirements for letting us realize this venture (I already found a person setting up Asterisk, I just need an answer to my question).
You can answer in German or English.
Many thanks for your time and help,
Chris Haywood
_______________________________________________________________
Hallo,
vielen Dank fuer Ihr Interesse in dieses Post. Unsere Anwaltskanzlei in Frankfurt mit mehrere Bueros moechte eine Beratung via Telephon anbieten.
Der Kunde ruft eine Mehrwertnummer an und kommt in eine IVR wo er die Nachrichten von die Anwaelte anhoeren kann und muss dann seine Wunschanwalt mit 3 stelligen Code eingeben. Dann wird der Kunde zu der Antwalt die er eingegeben hat weiterverbunden (jeder Anwalt hat ein eigene analog Telefonanschluss mit Nummer)
Diese Loesung soll auf Voip funktionieren mit Asterisk Software die die Anrufe weiterleiten wird. Die Mehrwertnummer wird den Anruf von die Kunde weiterleiten mit eine SID Protokoll auf die VoIP Server Asterisk (Anbieter CNS24 hat gesagt dies seie moeglich).
Nun zu der Frage: Was fuer ein Software (oder Hardware) muss eingesetzt warden dass die Anrufe die empfangen sind von die VoiP Server (Asterisk) zu die Anwaelte weitergeleitet wird auf die analoge Telefone (PSTN)? Brauchen wir eine speziellen SID Provider oder ein spezielles Voip Gateway?
Ich bin leider kein telecommunication experte, aber ich wurde es sehr schaetzen wenn Sie mir die Anforderungen fuer dieses Vorgehen sagen wuerden?
Entschuldigen Sie bitte das Deutsch (es ist die Uebersetzung von oben).
Vielen Dank fuer Ihre Hilfe die in English oder Deutsch sein kann,
Chris Haywood
first of all thank you for your time and reading this.
This is my first post in this forum and I hope you don’t mind that I post in English. (I’ve been living in Germany for just 5 months but I don’t have difficulties understanding German).
We are a team comprised of 52 attorneys at law based in Frankfurt (in different offices though) and we specialize in advising international clients on various legal issues affecting German law (ranging from criminal and private law, banking to real estate and much more).
Right now we have a lot of clients calling from all over Germany who like to take advantage of our services without traveling to Frankfurt. Hence, I came up with an interesting solution:
Legal counseling via premium rate numbers and VoIP!
I will be setting up German premium rate numbers (0900 and 0180 for prepaid where the fee per minute can be flexibly changed) which our customers can conveniently call. The caller is then introduced to an IVR application, introducing all lawyers (and their specialization) that are logged in to the caller by playing their prerecorded welcome messages. After all welcome messages have been played; the caller is prompted to choose his or her desired lawyer by entering the lawyer’s 3 digit extensions via phone. Alternatively, the caller may also skip the welcome message by directly entering the extension of his or her desired lawyer.
However this application should NOT rely on an ISND/Primary Rate Interface solution (also commonly known as T1 in the US), but rather on a Voice over IP solution powered by Asterisk. The incoming calls from the premium rate numbers would be forwarded via VoIP using the SID protocol, thus allowing us to lower the costs significantly. That’s why I’d like to setup a dedicated Linux server, running Asterisk on it.
I already spoke to our premium rate service number provider (CNS24 AG) and they assured me that they could forward (or route) all incoming calls to my VoiP server (Asterisk) using the SID protocol.
But now to my question: What kind of software (or hardware) do I need to forward the received calls from the Voip Server (Asterisk) to our lawyers which are located in different offices and still use analogue phones (over PSTN)? Do I need a special SID provider or a special Voip Gateway?
Unfortunately I’m not a telecommunication expert, but I’d appreciate if you could tell me the requirements for letting us realize this venture (I already found a person setting up Asterisk, I just need an answer to my question).
You can answer in German or English.
Many thanks for your time and help,
Chris Haywood
_______________________________________________________________
Hallo,
vielen Dank fuer Ihr Interesse in dieses Post. Unsere Anwaltskanzlei in Frankfurt mit mehrere Bueros moechte eine Beratung via Telephon anbieten.
Der Kunde ruft eine Mehrwertnummer an und kommt in eine IVR wo er die Nachrichten von die Anwaelte anhoeren kann und muss dann seine Wunschanwalt mit 3 stelligen Code eingeben. Dann wird der Kunde zu der Antwalt die er eingegeben hat weiterverbunden (jeder Anwalt hat ein eigene analog Telefonanschluss mit Nummer)
Diese Loesung soll auf Voip funktionieren mit Asterisk Software die die Anrufe weiterleiten wird. Die Mehrwertnummer wird den Anruf von die Kunde weiterleiten mit eine SID Protokoll auf die VoIP Server Asterisk (Anbieter CNS24 hat gesagt dies seie moeglich).
Nun zu der Frage: Was fuer ein Software (oder Hardware) muss eingesetzt warden dass die Anrufe die empfangen sind von die VoiP Server (Asterisk) zu die Anwaelte weitergeleitet wird auf die analoge Telefone (PSTN)? Brauchen wir eine speziellen SID Provider oder ein spezielles Voip Gateway?
Ich bin leider kein telecommunication experte, aber ich wurde es sehr schaetzen wenn Sie mir die Anforderungen fuer dieses Vorgehen sagen wuerden?
Entschuldigen Sie bitte das Deutsch (es ist die Uebersetzung von oben).
Vielen Dank fuer Ihre Hilfe die in English oder Deutsch sein kann,
Chris Haywood