[asterisk-biz] Remote SIP monitor

[Alex Balashov]

If you want more sophisticated waveforms than DTMF, you may want to look 
at Kristian Kielhofner's 'recqual', then.

On 01/06/2010 04:15 PM, Erik Lagerway wrote:

> Yep, was think about something along those lines in conjunction with
> what I mentioned earlier wrt SIP traces etc.
>
> Were they using their own SIP endpoints to gather that data or was there
> heavy lifting (adding an appliance, server etc.) in the network required?
>
> On Wed, Jan 6, 2010 at 1:05 PM, Lenz Emilitri <lenz.loway at gmail.com
> <mailto:lenz.loway at gmail.com>> wrote:
>
>     I think I spoke to a company in Finland that does something like
>     that, but instead of sending DTMF they send a predefined file
>     containing actual voice samples and reference tones.
>
>     They have a piece of software that analyzes what comes back,
>     compares it to the reference source and computes a number of
>     audiometric measures, like MOS estimators plus the obvious packet
>     loss, jitter, etc.
>
>     It sounded interesting but never had the time to test it in production.
>     l.
>
>
>     2010/1/6 Alex Balashov <abalashov at evaristesys.com
>     <mailto:abalashov at evaristesys.com>>
>
>         One thing we've done for a couple customers in the past is write a
>         script that initiates a call (via AMI Originate command) out of a
>         termination provider, which loops back into an origination
>         provider and
>         is received by the same Asterisk instance.  Once the call is
>         established, DTMF digits are passed and verified received in both
>         directions.
>
>         If this fails to take place or if the incorrect or incomplete digit
>         sequence is received, an SNMP trap was thrown via System().
>
>
>
>     --
>     Loway - home of QueueMetrics - http://queuemetrics.com
>
>
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-- 
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671