Hello all !

I would like to setup asterisk in a way that the codec used between the asterisk server an the external sip-account provider (for example sipgate.de) is always the same (bandwidth optimized), and that the codec between asterisk and my softphone could by anything that my softphone supports.
So that if a call comes in from the sip-trunk, the codec is choosen between the sip-account provider and the asterisk server for best bandwidth, and not by my softphone which is limited in codecs and always forces a-law.

Thanks a lot for any help !

Regards,
Fabianus