[asterisk-biz] Remote SIP monitor

[Erik Lagerway]

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> 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>
>
> 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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