Choppy Call Recoding

Hi All, I have the most strange issue with call recording.

We have a distributed system, ie, the asterisk server is in a central location (hosted) with extensions registering via broadband etc. Office A, B, C etc type thing.

Office A, and B, have perfect call recording - with may be 1 or two dropped packets (gaps) in the WAV file over an average of 20-30 minutes. However office C has terrible call recordings - with exactly the same settings.

I’m turning to the community on this one because i simply don’t understand how this could be the case and don’t really know what direction to try.

  • I’m recording using MixMonitor (i understand this to be the correct method)
  • All calls (both sides) are in g729, pass-through. (handset - asterisk - upstream provider)
  • WAV file is as wav49 (WAV with GSM compression)
  • Disk is RAID - Never reported a disk queue over 1 ever! ( I even tried a ram drive - same thing, its not the disk IO)
  • Sound over the actual phone is almost perfect, agents dont report poor quality, or gaps, however the recordings are full of gaps.
  • Sounds like: Hello, y…u are sp…ing with Sa… how c…n i hel… you.
    (if it was an actual conversation it would not be possible to understand each other)
  • At most there are 5 calls being recorded at one time, although the choppy audio can happen when there are no other calls being recoded.

I’m pretty sure it not a resource problem, since on the same system and the same time its fine for others.

If it was say a jitter or line quality problem from the broadband connection, why then are both sides of the conversation choppy - the agent is as broken up as the customer.

Does anyone have any ideas? I have been scratching my head, and upgrading the box for months.

So ill give some feedback for anyone following.

The Office C (the problematic one) have started using a dedicated ADSL line for the voice, rather than trying to push the calls and the data over one line. (Although they said they had sufficient bandwidth)

So far the results look good?? Based on these preliminary results - it seems that the lost packets where effecting both sides of the conversation in the recorded file… im not entirely sure how that works?? im guessing it something to do with the Math of the mixing… something like 0*X always = zero… in this case if there is a lost packet on channel A, and you mix channel A with B, there will be zero bytes in the AB mixed channel.

Well that’s my theory for now.


Are you get any solution about your query ? Please check your Asterisk Server RAM, it is sufficient ?

Please mention that which version of Asterisk are you using.