The biggest hurdle is that it doesn't work. Ok - it does in general, but 1) you need a T.38-compatible termination (to get past ~90% reliability) 2) Whatever you do, you're unlikely to ever reach 99%. Unless you peer directly to the SIP termination network, standard packet drops and jitter will give you enough random connection errors to annoy at least one customer.<p>Other notes: 1) Please don't assume "voip == asterisk". Have a look at FreeSwitch or Yate too. 2) If you don't feel like reading and understanding most of RFCs 3261, 4566 and some texts about T.38 renegotiation -- start looking for some experienced VoIP person to set you up with the basic gateway. VoIP is unfortunately far away from the "make install && forget" types of services - it needs a maintainer who knows the stuff inside out.<p>So the best way to go about it? Unless you've got some VoIP / telecom. experience, I'd recommend finding someone who can do it for you either on the *-biz mailing lists, or on typical freelancer portals.