Gents, running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a have been fighting with this for an hour or more...have done a bit of research on the list and Google itself, scoured the Wiki etc. but can't seem to figure out where to set the codecs to be "well recd" by the remote SIP peer. This used to work fine until I upgraded 2 days ago and it stopped working without my doing / changing anything. The remote SIP peer is {*} which doesn't like multiple m= lines with differing ptime values. Debug on near-end says: "Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]" Full SDP here: v=0 o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80 s=FreeSWITCH c=IN IP4 192.168.1.80 t=0 0 m=audio 28884 RTP/AVP 98 8 3 101 13 a=rtpmap:98 L16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=audio 28884 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv Far end says: Rejecting non-primary audio stream: audio 28884 RTP/AVP 0 101 13 I have tried to play around with the codec globals in vars.xml, to no avail. have also added stuff directly in the dialplan like so: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="absolute_codec_string=PCMU@40i"/> <action application="set" data="sdp_m_per_ptime=false"/> <action application="bridge" data="sofia/gateway/mycore1/1111"/> but no dice. Tried to remove the "absolute_codec_string", still no dice. I read about the "sdp_m_per_ptime" that lets me "cheat" but that ain't doing anything either. What gives? Thx in advance