i am trying to find out if our idea of integrating asterisk in our Voice Plattform (such a system everbody knows from hotlines where you have automated dialogs using voice-recognition and text2speach).
We would like to reuse our Digium TE110P-ISDN card which allows us to use the 30 speech-channels coming from our E1/PRI-multiplexer.
The chain is supposed to look as follows:
A caller <-> E1/PRI-ISDN <-> 30-channel-Multiplexer <-> Digium TE110P <-> zapata/zaptel <-> asterisk <-> capi4Linux <-> ATIP Voice Server
The problem which lead us to this idea is that the voice plattform software “ATIP Voice Server” can not use zaptel to control the TE110P-card. ATIP only knows capi2.
The TE110P-card requires zaptel in order to access it.
So we had the idea of using asterisk as a “translator” from zaptel to capi4linux and vice versa. Does anyone knows if this is feasable with asterisk, that means, can asterisk access a TE110-card via zaptel and take the voice data from the 30 channels and “route” the voice channel data to a capi4linux-application like our ATIP-Voice Server (and the way back as well of course)?
Your help would be very appreciated. If i find anything out in the meantime, i will post it here!