I am trying to use meetme() on SIP channels. I found this line on voip-info.org
------------- It *is* necessary either to have a Digium card or a dummy timing driver (e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that doesn't help you use AGI with SIP channels: They have no capacity to use any AGI script at all. If they try to, they get no audio. ----------------- I am using 1.4.18 and I am hearing no audio. just like it says. I do have a digium TDm800 card in my box. Does asterisk 1.6beta have this same issue still???? How do people get around this issue? I am looking for a meetme conferencing in a handful of phones, call AGI to play a wave file then hangup.... My meetme is a pure SIP meetme. Thanks, Jerry _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
