[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed

[Matthew Roth]

>>On the other hand, in the case of Monitor()'ed calls, for ideal call  
>>recording quality, you'd want a jitterbuffer somewhere between the  
>>packets being received and being written to disk, but asterisk  doesn't 
>>have that.   You could do that by enabling the jb for VoIP- >VoIP calls 
>>when Monitor() is active, which would add latency to  these calls, or some 
>>other way, which would require some more code to  implement.
>
>Using the new MixMonitor infrastructure, it would be relatively easy to put 
>a jitterbuffer in between the frames being copied from the channels and 
>them being mixed/written.

Just a couple of questions about MixMonitor:

- What is the overhead of mixing the legs locally via MixMonitor?  Currently 
we're moving the legs to a remote machine for mixing to maximize the CPU 
available to Asterisk.

- Can digital recording via MixMonitor be triggered from agents.conf and 
queues.conf as well as extensions.conf (as it can via Monitor)?

- What's the ETA on MixMonitor being introduced to Asterisk Business 
Edition?  If it's a while out, can it simply be added to our current 
installation as a new module?  We did this at one time with the MySQL CDR 
module.

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer