I think what Todd was referring to was to JUST do the signaling proxy on the Asterisk but not proxying the media.
This is the definition of a SIP proxy. Asterisk is a PBX that supports SIP, but not really a SIP proxy. As a PBX, it wants to be in the middle of a call. As an additional feature, it can release the media stream to the SIP devices if these are on the same network, either outside or inside of a NAT, but not on two sides. And this only works if the device supports a SIP re-Invite. http://www.voip-info.org/tiki-index.php?page=Asterisk+sip%20canreinvite
Use SIP Express router from iptel.org if you want an excellent open source SIP proxy.
as well. I was thinking of doing the same thing if possible with H323, ICan't help you here, Jeremy?
/O
_______________________________________________ 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
