I am trying to test mod_silk with FreeSWITCH rev. 17118. The problem appears to be that I haven't been able to get my FreeSWITCH to offer SILK as one of its codecs. mod_silk is loaded. I am using the PortAudio interface to make the call; the other end is running a recently built FreeSWITCH as well, thanks to Bill Acker for participating in the testing as always. I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with SILK@24000h as the first item in the comma-separated list. I have also tried it with SILK (no sampling rate specified). The SIP trace doesn't show that it is being offered. Bill has configured the other end in the same way. For example: a=rtpmap:10 L16/48000 a=rtpmap:114 CELT/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 is what appears as the SIP message from my local FreeSWITCH with SILK@24000h,CELT@48000h,G722,BV32,PCMU,PCMA,GSM in the codec preferences. (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.) Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and IPv6 for making the connection. I know Brian and others have this working, so there's obviously something amiss here. As usual, there's no urgency - it's another attempt at a community contribution through testing.