> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Olle E. Johansson > > Brian Cuthie wrote: > > > > 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. > Asterisk SIP does not support silence suppression. In fact, > using Silence suppression on an inbound RTP stream will lead > to problems, since Asterisk takes timing from inbound RTP streams. >
Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users