[asterisk-biz] Remote SIP monitor

[Lenz Emilitri]

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>

> 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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