[asterisk-dev] Silence detection on Dial application (mainly with SIP)

[Nir Simionovich]

Hi All,

  Here's an interesting feature that can be added to app_dial. I hadn't
written it (have no idea where
to start with it) - but I believe that it's worth spending some time
figuring this one out.

  On callback based systems, I've noticed that their exists an issue with
disconnecting the call when
there is silence on the line. Asterisk is fairly capable of detecting the
absence of RTP on SIP, however,
if RTP is still existant between the nodes, Asterisk doesn't disconnect -
although the RTP is just silence.

  Is there a way to add an ability to app_dial to do this?

Nir
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