Hi, Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which requires transcoding from g.722 (or other codec which declares 8kHz sampling rate in the SDP) to SPEEX/16000 or SPEEX/32000. If the calling extension uses only g.722, alaw, ulaw, etc, then only the SPEEX/8000 narrowband variety of SPEEX is offered to the recipient extension in the SDP of the SIP invite. If the call is initiated the other way round - e.g. Client using SPEEX/32000 --> FreeSWITCH --> Client using g.722, then the call is transcoded with no problem. I am wondering if this is the intended behaviour, to avoid transcoding narrowband --> wideband. However what I am finding is that transcodinig g722/8000 (wideband) to SPEEX (wideband or ultrawideband) does not seem to work. I would be most grateful if anyone were able to let me know if there is a configuration option I can set to alter this behavious and allow the full range of SPEEX sampling rates to be offered in the SDP to the receiving party, regardless of the codec used by the calling party. Also, is this perhaps different in a more recent version? Many thanks Bruce