I've created the following really simple dialplan:
;;;exten => 100,1,Monitor(gsm,ast_mon_${TIMESTAMP})
exten => 100,1,Monitor(WAV,ast_mon_${TIMESTAMP})
exten => 100,2,Dial(SIP/jmjones-l1,20,Ttr)
exten => 100,3,StopMonitor( )
exten => 100,4,VoiceMail(u100)
exten => 100,5,Hangup()
I've alternated between gsm, WAV and wav and have encountered the same
results: the "live" audio between the calling parties is good....good
enough for what I want to accomplish, but the recorded audio quality is
a little choppy.
Basically, what I'm wanting is for one person to be able to call another
person via SIP and have asterisk handle the conversation so it can be
recorded. Both ends of the conversation are using gsm and as I
mentioned, I've tried recording to a gsm file as well as wav and WAV.
During the call, CPU isn't pegged, nor is disk IO, nor is network IO.
Has anyone seen anything like this? Anyone have any alternatives they'd
like to propose?
BTW, I'm running Asterisk 1.0.9 on Ubuntu Breezy and have run Linphone
on Linux, SJPhone on Linux, and SJPhone on Windows all with the same result.
Any help is greatly appreciated.
- Jeremy Jones