[Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?

[Brian Cuthie]

Let's say that I have a call coming in to Asterisk through a TDM400P and
going out through SIP to someone on the Internet. Is there any configuration
option that would allow me to do silence suppression on the RTP stream
generated by Asterisk on behalf of the TDM400P connected user?  SIP phones
allow me to do this easily, but I'd like to be able to conserve upstream
bandwidth on calls that don't emanate from a SIP phone here at my location.

Thanks

-brian
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