[asterisk-dev] Voice format when using another SIP server

[Jamel A]

Hello,

I'm using Asterisk and Sphinx4 to do live voice recognition. I convert all
the voice stream to PCM format so Sphinx could decode.
Everything works very well.

My problem, is that I've added to sip.conf another SIP server. So I suppose,
that all the input voice stream are transmitted from this latter to Asterisk
(itself transmitting to Sphinx4). But unfortunalty the voice recognition
doesn't works.

So my question is, do Asterisk convert anyway to the format that I've
indicated? or does it keep the sip server format even if I tell it to
convert to PCM format?

Thanks
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