I'm experiencing a strange problem on a recording of an audio stream where the two channels (I am recording in stereo) are not in sync. By that, I mean when I play back the audio one of the tracks is somehow time-shifted, so that it doesn't match up to the proper place in the timeline with the other side. For instance, if I call out to a bank IVR system that plays a prompt, and I barge into the prompt by pressing a DTMF key, on the recording I will hear the prompt start and then stop at the same place it stopped when I pressed the DTMF, but the DTMF key itself does not show up on the recording for 5 seconds more. Has anyone ever heard of anything like this? Now, a few more details. I am creating a bridge conference and then I am recording the audio stream to and from the called party and the conference. I do not have this problem if I record the audio stream to and from the calling party and the conference. Here is the dial plan I am using: <extension name="6174973895" continue="false"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="6174973895"> <action application="ring_ready"/> <action application="answer"/> <action application="start_dtmf"/> <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> <action application="export" data="outfile=$${base_dir}/tmp/${call_id}.wav"/> <action application="lua" data="demo_call2.lua"/> <action application="export" data="RECORD_STEREO=true"/> <action application="export" data="nolocal:execute_on_media=record_session ${outfile}"/> <action application="export" data="nolocal:api_on_answer=conference ${call_id} play misc/this-call-may-be-recorded.wav async"/> <action application="conference" data="bridge: ${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/> </condition> </extension> Any thoughts on what might be going on?