[asterisk-biz] Remote SIP monitor

[Erik Lagerway]

K, couldn't find what I wanted so I built it. It's lacking the MOS scoring
and media events but does watch for dropped calls and dropped SIP endpoints.
It's been monitoring my network for a month now and works pretty well
although I have only tested a few servers.

Check out here, it's free to try.. http://sipqos.com

Feedback always welcome.. good, bad, indifferent.

- Erik


On Thu, Jan 7, 2010 at 7:05 AM, <lists at contacttel.com> wrote:

> Done that as well to test PRI's, but then on some small network congestion
> peeks it would give false positives and reboot boxes that were healthy.
>
> Not sure but i guess it's still a good option for that part, as for the
> rest, asterisk needs a monitor app built-in, as all other options  are
> either not powerful enough or too much (requiring a steep learning curve or
> config time)
>
> >>-----Original Message-----
> >>From: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz-
> >>bounces at lists.digium.com] On Behalf Of Alex Balashov
> >>Sent: January-06-10 2:29 PM
> >>To: asterisk-biz at lists.digium.com
> >>Subject: Re: [asterisk-biz] Remote SIP monitor
> >>
> >>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().
> >>
> >>--
> >>Alex Balashov - Principal
> >>Evariste Systems
> >>Web     : http://www.evaristesys.com/
> >>Tel     : (+1) (678) 954-0670
> >>Direct  : (+1) (678) 954-0671
> >>
> >>_______________________________________________
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>
>
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