Hallo,
sorry my German is not sufficient enough to write understandable questions. If anyone wants to answer in German, that is OK, since I understand 90% of it...
I'm very happy with my FBF so far, connecting to adsl, having 3 incomming/outgoing MSN isdn numbers (FON 1 to 3), and one "internetrufnummer"...
I've also tried to use the new inbuilt "anrufbeatworter" with success.
I NOW WANT TO GO ONE STEP FURTHER:
I want the anrufbeantworter to pick up the phone on one isdn MSN number, than play an "ansage" and than perform a "rufumleitung" to an external number, either using the ougoing isdn line, or using the sip connection.
QUESTIONS:
1/ Do I need asterisk to do this? Or, can the current default "anrufbeatworter" be hacked to do this?
2/ If I need asterisk, how to proceed? I've been reading most of the forum last couple of days, struggled a bit to get * installed, since the missing libresolv.so.0 library, but this can be worked around. I guess I need to edit /var/tmp/asterisk/extensions.conf, /var/tmp/asterisk/sip.conf, and make my own voicemail.conf? But I have to admit I'm a bit lost here... how to proceed?
Thanks for any advice...
Stefan.
sorry my German is not sufficient enough to write understandable questions. If anyone wants to answer in German, that is OK, since I understand 90% of it...
I'm very happy with my FBF so far, connecting to adsl, having 3 incomming/outgoing MSN isdn numbers (FON 1 to 3), and one "internetrufnummer"...
I've also tried to use the new inbuilt "anrufbeatworter" with success.
I NOW WANT TO GO ONE STEP FURTHER:
I want the anrufbeantworter to pick up the phone on one isdn MSN number, than play an "ansage" and than perform a "rufumleitung" to an external number, either using the ougoing isdn line, or using the sip connection.
QUESTIONS:
1/ Do I need asterisk to do this? Or, can the current default "anrufbeatworter" be hacked to do this?
2/ If I need asterisk, how to proceed? I've been reading most of the forum last couple of days, struggled a bit to get * installed, since the missing libresolv.so.0 library, but this can be worked around. I guess I need to edit /var/tmp/asterisk/extensions.conf, /var/tmp/asterisk/sip.conf, and make my own voicemail.conf? But I have to admit I'm a bit lost here... how to proceed?
Thanks for any advice...
Stefan.