I have an application where I create a bridging conference that outdials a B party, and then I record the stream between the conference and the B party. However, the two channels of the resulting stereo recording are not in sync: for instance, if I call into an IVR and barge into a prompt by pressing a dtmf, on the recording the dtmf tone appears several seconds after the barged-in prompt stopped. Its as if the stream from the conference to the B leg has been shifted in time so it no longer matches up properly with the conference mix. I can't tell if this variance is constant or gets worse as the conference goes on. Has anyone a notion of what might be going on here? Here is my dialplan: <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>