Asterix 1.2beta ignores rtp.conf


1.2beta seems to ignore the rtp.conf file. During startup I see that rtp.conf ist read and ports from 10000->20000 will be used.

However if I make a sip call while rtp debug is on, I see that it will try to connect to the callee using a port number around 40000, which will be blocked by my firewall and therefore no audio can be heared by the callee.

Any hints?