[asterisk-users] asterisk canreinvite=yes

[satish patel]

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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