[Freeswitch-users] stereo recording problem - tracks out of sync

[Dave Horton]

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>