On Tue, 2010-02-23 at 08:22 -0500, Michelle Dupuis wrote: > We're creating a SIP gateway for a client that will take one leg of a > call in via SIP, and out the other side via H.323. To minimize load > on the gateway, we would like to have the RTP stream bypass the > gatewayy altogether (directrtp/reinvite). Is this possible with these > to protocols? > > Thanks > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
IMHO, It's impossible ;) -- Best regards, Vince Mallow xmpp: [email protected] web: http://gentoo-way.blogspot.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
