Dear forum!
Disclaimer: I’m a VoIP newbe, so please forgive me if my question is stupid. I hope it makes sense (at least).
AFAIK (doesn’t say so much) EuroISDN uses G.711 A-law as audio codec for voice calls, and most SIP-terminaters know G.711, too. Is it possible to configure asterisk (w/ ISDN-adapter) that it passes through the G.711-packets without decoding/coding/altering? It shall just change the encapsulation from ISDN to RTP and vice versa.
p.