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<TITLE>Silence suppression on SIP calls generated from Asterisk?</TITLE>
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<P><FONT SIZE=2 FACE="Arial">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.</FONT></P>
<P><FONT SIZE=2 FACE="Arial">Thanks</FONT>
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<P><FONT SIZE=2 FACE="Arial">-brian</FONT>
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