[asterisk-users] SIP Quality Metrics
I've been reading about ${RTPAUDIOQOS}, which supposedly contains call quality metrics, but I can't seem to convince Asterisk to grab it for me after the call. Anybody have any luck with it? Anybody know how to get Asterisk to executive a command after the call is over? Thanks, Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Quality Metrics
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