Hello, I am using a GSM based endpoint connected to freeswitch that makes calls to the PSTN via a SIP gateway (SBC). The SBC uses PCMU between itself and freeswitch. When I make an outgoing call from a GSM based device via freewsitch to the PSTN via the SBC, everything works fine and audio works in both directions for both end points. I looked at the console logs and they do indicate that I am using GSM. Console output when I dial and before answer on the GSM device: v=0 o=- 74 0 IN IP4 10.229.0.58 s=session c=IN IP4 10.229.0.58 b=CT:17 t=0 0 m=audio 59806 RTP/AVP 8 0 3 97 101 k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:optional Console output once it rings and after I answer on the PSTN side: v=0 o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10 s=FreeSWITCH c=IN IP4 10.229.0.10 t=0 0 a=sendrecv m=audio 30896 RTP/AVP 3 101 13 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 When I receive a call from the SIP gateway, the endpoint making the call (not on freeswitch) can't hear me speaking from the GSM device connected to freeswitch. I can hear everything fine on the GSM device. Here is the console output for the call info coming in from the PSTN. v=0 o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132 s=FreeSWITCH c=IN IP4 38.113.164.132 t=0 0 a=sendrecv m=audio 16724 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Here is how I have vars.xml configured: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> When I prioritize GSM on the outbound codec prefs I get static on the PSTN side. <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/> Any ideas? Maxim.