Join #freeswitch on freenode... People are there right now that can help you On 8/10/11 3:29 AM, "Viktor Jamrich" <viktor_jamrich@centrum.sk> wrote:
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.
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