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.
HopeThisHelps
2006/5/22, Pieter Claassen <[EMAIL PROTECTED]>:
I find that when asterisk answers the phone, the initial second or so is lost.
I can imagine that echotraining can do this, but this is between SIP phones
and I don't think there is any echotraining enabled?
BTW. Asterisk is definitely playing sounds that first second (The CLI would
indicate that it would play a beep but I just won't hear it).
Any comments appreciated.
Pieter
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