[Asterisk-Dev] Record app on Sip channel doesn't send RTP

[Kevin P. Fleming]

Tony Mountifield wrote:

> I have found, using Ethereal, that while Record is executing, no outgoing
> RTP frames are generated. Is this correct behaviour? I would have expected
> an RTP stream containing silence, since my understanding is that Asterisk
> does not yet support silence suppression.

This is a known issue and is currently open in the bug tracker... we 
just haven't settled on a solution yet.