I am trying to use meetme() on SIP channels.
I found this line on voip-info.org
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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.
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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
