[Freeswitch-users] transcription module development

There are a bunch of cloud-based transcription services these days (Nuance, google, watson etc) that provide voice to text, and I am working on developing a freeswitch integration. It seems to me that they don’t fit quite precisely into the mod_asr framework as there is, for instance, no concept of a speech grammar. Just wondering from the freeswitch devs, whether they would recommend that this be implemented as an asr module anyways (i.e stub out and ignore grammar) or should it really be a new type of module.