[Asterisk-Users] Initial second lost on SIP phones

[picciuX]

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 <pieter at claassen.co.uk>:
>
> 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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