[asterisk-users] Detect Low Quality Calls - Realtime

[Robert-GMAIL]

Asterisk "sip show peers" lists the qualify value in ms (milliseconds).

Please read up on this and the setting for it in sip.conf config file

Sent from my iPhone 5

On Jan 5, 2013, at 5:30 AM, XBrian <boboodz at yahoo.co.uk> wrote:

> Joachim, thanks for the reply
> - delay you can somewhat estimate prior to the call (with qualify for example)
>>>> Pls be explicit. How do I use qualify to measure delay
> 
> -  The jitter / packetloss you can only figure out when the call is already 
> up for a while. 
>>> what would you use to measure jitter / packetloss in real time?
> 
> 
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users