[Freeswitch-dev] Speaker Identification Implementation

[Viktor Jamrich]

Hi guys,
our company wants to use speaker identification/verification (SID) in 
FreeSWITCH. However, only ASR and TTS are currently implemented
so we decided to contribute by coding the SID interface as well as 
speaker identification/verification part in mod_mrcp.
Is it possible to get our code commited and made available to public 
within a month ??
And what's the procedure in getting our code commited ??
I think there's a huge gap on the FS web in information for newbies 
wanting to help with coding...

PS: I'm not sure whether this is the right place to ask such question 
but I've been to #asterisk-dev on IRC and nobody answered my question...

Thanks for help.