----- Original Message ----- From: "Jeremy McNamara" To: <[EMAIL PROTECTED]> Sent: Monday, May 03, 2004 12:48 PM Subject: Re: [Asterisk-Users] Asterisk remains in the media path
> brian wrote: > > >Can't do it because you are changing from one technology to another. > > > > > > > > Actually its cuz chan_h323 sucks like that. > > > Jeremy McNamara So does this mean you could get direct RTP steams between a SIP client and a IAX2 client? What about inband/out of band DTMF issues? Thanks, Jim > > >>-----Original Message----- > >>From: [EMAIL PROTECTED] [mailto:asterisk-users- > >>[EMAIL PROTECTED] On Behalf Of Paul Berger > >>Sent: Monday, May 03, 2004 10:29 AM > >>To: Liste Asterisk > >>Subject: [Asterisk-Users] Asterisk remains in the media path > >> > >>Hi all, > >>Just a quick question: I have an H323 terminal and some MGCP phones > >>connected to *, and when they call each other * remains in the media > >>path no matter what (while I'd like to have the RTP stream directly > >>between the phones). > >>- mgcp.conf has canreinvite=yes > >>- extension.conf doesn't contain any Dial() instance with t or T > >>Did I forget something? > >>Thanks, > >>Paul _______________________________________________ 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
