[Asterisk-Users] canreinvite=no and codecs.

[Nikolay Pavlov]

Hi, folks.

If i use canreivite=no option in my sip.conf for users is this mean that 
i need to load 729 and 723 codecs for thos UA that want to transmit it?
Or this is just traffic redirection feature? How this option reflect on
server load?


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= Best regards, Nikolay Pavlov. <<<--------------------------------------- =
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