Hi, On Fryday, 2007-02-02 François Delawarde wrote : > Don't you think it could be an interesting feature in Asterisk? It > already does transcoding, why not gain when voice flow passes through it? > > François.
On a "SIP-to-SIP"-Call Asterisk is not neccessarily in the voice flow, so this does not work in any case. Karsten _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
