Hi everyone,
I haven't used Asterisk in many years, but in searching for a good
podcasting solution that will allow me to record three or four
participants to individual tracks (which would allow me to go in and
do noise removal on each participant individually, giving a higher
quality), I came up with the idea to use Asterisk. Now I've installed
it and got it all set up and did a test call, and it mostly works, but
there's an issue. First, here's my dialplan section:
exten => 15,1,Answer
exten => 15,n,Monitor(wav16,,o)
exten => 15,n,ConfBridge(10)
As you can see, it's pretty simple. It begins monitoring the caller's
audio only, and dumps them into the conference. I tested this with
myself and just one other person. I line up the starts of the WAV
files, and it's fine at first. But after about 5 minutes, the audio
goes out of sync, and one of us is ahead of the other, and it gets
worse throughout the call, even if I split the WAV and realign every 5
minutes, it's still off.
All of us are using IAX2 via Kiax (I on Linux, the others on Windows),
all using the ULAW codec. There's nothing else going on with this
setup. The system specs (Dual Core 3GHz, 3GB RAM) should be able to
handle recording to 3 or 4 WAV files at once, I'd imagine, so I don't
think that's the issue.
Is there an alternate way to record all participants of a ConfBridge
or MeetMe or something, but not mixed, to individual files, AND have
them line up correctly? If I am on the right track, is there something
I am missing that I could try to fix the issue? I was thinking maybe
it had to do with jitterbuffer or something (I believe it's disabled
by default), but I was reading a bunch of info on the web to try and
figure this out, and I forget where but I read something about the
jitterbuffer being unsupported in MeetMe, Record, etc apps, but might
be enabled using the Dial command and Local channel?
If anyone can point me in the right direction, I'd appreciate it.
Thanks!
Dana
