[Asterisk-Users] Where to buy POLYCOM phones (forcing native bridge between SIP terminals)

[Tom Schroer]

> 
> Message: 2
> Date: Mon, 18 Oct 2004 16:08:30 -0400
> From: "M. Willigs" <mwilligs at conexiongroup.com>
> Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 	<asterisk-users at lists.digium.com>
> Message-ID: <005101c4b54e$3f380910$ca0aa8c0 at 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
>