Hello world,

I've got two FS servers configured as gateways for each other and I'm currently testing the telephony. Usinge the ILBC audio codec, I figured out that one of the FS servers doesn't forward RTP streams correctly to the other server. Here is its status-quo:
INPUT = proper ILBC payload type (97 or 108)
OUTPUT = unknown payload type (97 or 102)

I've already changed the parameters in internal.xml & external.xml:
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="disable-transcoding" value="true"/>

When dialing out, I also use the following syntax:{absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber

Is there another thing to do to have proper ILBC streams passing through the gateways ?
Thanking y'all in advance ;)

BR,
David N.