Podcasting with Asterisk - monitor files don't match up

My goal is to have a separate “microphone” or “input” recording for each person on a conference call. The reason I want to isolate each voice is so I have more granularity in applying effects and amplitude on a final master recording of all voices.

I created a simple extension that calls into a meetme conference room and records the input and output files using Monitor.

[quote]exten => 300,1,SetVar(CALLFILENAME=${CALLERID(num)}-${TIMESTAMP})
exten => 300,2,Monitor(wav,${CALLFILENAME})
exten => 300,3,Meetme(10,c)
exten => 300,4,Playback(vm-goodbye)
exten => 300,5,Hangup

For each person that enters the conference room, I get another set of input/output files. The file naming convention is based on their originating extension number and a time stamp down to seconds. (milliseconds sure would be nice but I couldn’t find that variable!)

In theory I thought I should be a be able to pull each of the “input” recordings into a program like Audacity or Audition and line them up on a timeline based on their time stamp… but they won’t line up.

It’s about 2am here right now so its possible that my computer is playing tricks on me, but it even seems like the duration is different on the recordings.


Person A enters the conference room at 01:00AM and recites the alphabet through a kazoo. Then at 01:05AM Person B joins the conference and both exchange in a dialogue.

I pull Person A and B’s output WAV file into sound editing software of choice. I start Person A’s output file at 0:00:00 on the timeline and bump Person B’s output file to 0:05:00 (five minutes later). In theory the two output files should contain a coherent version of the conversation, but they don’t.

Anybody have a clue as to why this doesn’t work out? If I add buffers to the meetme.conf file will it fix this?