Has anyone played around with QoS or TOS relative to * and sip phones? I was just doing a little real-time research and noticed our C7960's mark IP packets with "low delay" and "high throughput" (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets flowing "from" asterisk back to the sip phone are not marked at all.
Is there a * config parameter to enable such a function? Rich _______________________________________________ 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
