[asterisk-users] Limit Asterisk

Ron Wheeler rwheeler at artifact-software.com
Wed Jul 23 16:01:49 CDT 2014

I would also do some math on the bandwidth requirement.

If you divide your disk bandwidth by your recording bit rate what is the 
theoretical maximum number of calls that you can record at once? Assumes 
that you have infinite CPU and memory and that you can actually drive 
the disks at their maximum.
If this comes out to 300, you are already there. If it comes out to 
3000, you have something wrong in your setup or your assumptions and a 
target to work towards.

What quality are you using in the recording? 44k per second(CD quality 
sound)  uses a lot more bandwidth than 3K (telephone quality)
What encoding are you using?
How low a bit rate can you use and still have usable recordings? If they 
are for legal or audit use, you can go pretty low. If you are recording 
soundtracks for reuse in training or publication, you may require higher 
bit rates.

If you disable recording, how many simultaneous calls can you support? 
Just to be sure that recording is the issue.


On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
> Your bottleneck is most likely your drive bandwidth.  Even with SAS 
> drives, you'll need to move to a raid 5+ solution with 6+ drives to 
> continue to increase the concurrent calls, or use a storage appliance.
> To confirm this, install the tool nmon and use the v and d options to 
> bring up the resource usage indicators and drive busy/throughput 
> statistics.
> On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones 
> <eduardo at ypytecnologia.com.br <mailto:eduardo at ypytecnologia.com.br>> 
> wrote:
>     people
>     I have a running Asterisk 1.8.28 in great Dell server with two
>     xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This
>     server is recording all calls (placed to record the audio in a ram
>     disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each
>     call runs some validation and AGI's have an auto dialer system
>     that generates calls over the manager. Calls originate and
>     terminate via SIP (no transcode).
>     With this structure, even being a great server, we can not spend
>     150 simultaneous calls. When it reaches 140, the load average goes
>     up a lot and the calls start to get very bad audio, tear, etc..
>     Using the top we see that all the processing is for asterisk. In
>     this scenario, I think there is some limitation in Asterisk, or
>     even the manager due to the auto dialer.
>     Can anyone give me any tips where I can look where is the
>     bottleneck? I need to get at least 250 calls that server quality.
>     tks
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> Scott Griepentrog
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Ron Wheeler
Artifact Software Inc
email: rwheeler at artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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