<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">Is it possible to host call waiting
service on Asterisk for a SIP device?</font>
<br><font size=2 face="sans-serif">What i am trying to achieve is that
while a SIP user is busy on a call and a new call for that user comes in,
asterisk should play the call waiting tone to that user.</font>
<br><font size=2 face="sans-serif">I have a vague idea that if i can get
hold of the existing bridged channel when a subsequent call is received,
i can then redirect that channel to play tone. </font>
<br><font size=2 face="sans-serif">The problem is how can i get hold of
the bridged channel in the first place? Also is there a better way of accomplish
call waiting.<br>
</font>
<br><font size=2 face="sans-serif">TIA,</font>
<br>
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