On Mon, 2005-04-11 at 16:40 +1000, Rod Bacon wrote: > I don't know if what you're trying to do is possible, but the easiest way to > check would be to take a look at the raw packets on the ethernet interface > of your * server once a call is in progress. If indeed the RTP can be handed > off to the 2 endpoints, you should only see SIP traffic at your server. > TCPDUMP is your friend.
or sip debug, or iptraf/jnettop/any other network traffic monitor. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 0000 [EMAIL PROTECTED] Fax: +61 2 9345 4396 www.websitemanagers.com.au _______________________________________________ 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
