<div>that second is time needed to establish the RTP session, AFAIK. Simply put in the dialplan a "Wait(1)" between the "Answer" and the "Playback". It solved same issue for me.</div>
<div>HopeThisHelps<br><br> </div>
<div><span class="gmail_quote">2006/5/22, Pieter Claassen <<a href="mailto:pieter@claassen.co.uk">pieter@claassen.co.uk</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I find that when asterisk answers the phone, the initial second or so is lost.<br>I can imagine that echotraining can do this, but this is between SIP phones
<br>and I don't think there is any echotraining enabled?<br><br>BTW. Asterisk is definitely playing sounds that first second (The CLI would<br>indicate that it would play a beep but I just won't hear it).<br><br>Any comments appreciated.
<br>Pieter<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
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