I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy.
I think what you are looking for is QOS (quality of service). There is a good wiki page (www.voip-info.org) on it. I personally use the wondershaper script.
-- Cheers,
Matt Riddell _______________________________________________
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