Dear all
I have small lan and i have configure hardphone with my
asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in
sip.conf
If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come
in media path
and if i user conreinvite=yes then RTP path would be sip phone to sip phone ???
My all phone in LAN not behind the NAT so guessest me what option would be best
for my setup
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Satish Patel
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