Tom Schroer
2004-Oct-18 13:02 UTC
[Asterisk-Users] Where to buy POLYCOM phones (forcing native bridge between SIP terminals)
> > Message: 2 > Date: Mon, 18 Oct 2004 16:08:30 -0400 > From: "M. Willigs" <mwilligs@conexiongroup.com> > Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <005101c4b54e$3f380910$ca0aa8c0@miguelpc> > 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 >