[asterisk-users] Call quality

Steve Totaro stotaro at totarotechnologies.com
Tue Jul 1 10:37:13 CDT 2008


Try IOSTAT http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat

Maybe you can correlate VM and/or emailing of VM to your IO spikes.

Have you watched top and the Asterisk CLI when someone hits the panic button?

Thanks,
Steve T

On Tue, Jul 1, 2008 at 11:17 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
> The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
> of all alerts are internal calls. I had the chance to notice the problem
> once myself but I could never again reproduce.
>
> Best regards,
> Loic Didelot.
>
> On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote:
>> On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
>> > Hello,
>> > one of my customers complained about bad voice quality on several calls,
>> > so I programmed a button on each phone which users can hit if they have
>> > audio drops and echo.
>> >
>> > I did this to check if there is a common recurrent problem to a given
>> > destination or just for one user etc... But till now I could not detect
>> > a pattern which could explain the problems
>> >
>> > This "alert button" is pressed between 7%-10% of all calls. The customer
>> > has 25 phones and around 300 calls per day.
>> >
>> > The SNOM phones are connected to Linksys switches and are totaly split
>> > from the computers network. The same goes for the asterisk box. No calls
>> > are routed trough the internet.
>> > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier
>>
>> Are the problems in SIP->PSTN calls? SIP->SIP calls?
>> PSTN->Local? (echo test, playback, whatever)
>>
>> SIP->PSTN or PSTN->SIP (what direction is the call)?
>>
>> 7% is something you have hope of reproducing. Unless you miss the real
>> factor. Have you managed to reproduce it yourself?
>>
>> >
>> > The carrier we use is known for his good quality and we never had a
>> > problem. It is the historic and most expensive carrier in Luxembourg.
>> >
>> > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
>> > maximum of 6 concurrent calls.
>> >
>> > Maybe someone can help me to track down the problem. What should I
>> > check, monitor test. Any ideas are welcome.
>>
> --
> Loïc DIDELOT
> MIXvoip S.a.
> ldidelot at mixvoip.com
> http://www.mixvoip.com
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list