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

[Tony Mountifield]

In article <4360EAC6.9070209 at digium.com>,
Kevin P. Fleming <kpfleming at digium.com> wrote:
> 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.

Ah, found it - #5135. Thanks!

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
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