[asterisk-users] SIP SDP keep original codec selection?

[Ian Hailey]

Hello all,

When an incomming SIP call is reveived I would like to force Asterisk to 
keep the SDP codec selection for the resulting outgoing call to the 
destination SIP endpoint. Does anyone know how this could be acheived? I 
know that the allowed codecs for each SIP endpoint can be restricted in 
the sip.conf but need this to be dynamic based.

Thanks

Ian.