> > Message: 2 > Date: Mon, 18 Oct 2004 16:08:30 -0400 > From: "M. Willigs" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi everybody. > > With te module oh323 SIP to SIP calls can be autorized in the > same * server in an h323 gatekeeper. I need to do it this way > because all the original system has been built under h323, > so, it's easy to integrate the Asterisk with the rest of the > system this way. > > My question is: is there a way to force a native bridge > between both SIP terminals in order to avoid the RTP trafic > across the Asterisk?
Not sure if this is what you are looking to do, however, have you tried setting canreinvite=yes in sip.conf? > > Thanks in advance.M. Willigs > _______________________________________________ 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
