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List: asterisk-biz
Subject: Re: [asterisk-biz] Remote SIP monitor
From: Erik Lagerway <erik () sipthat ! com>
Date: 2010-02-17 23:53:11
Message-ID: 4eace7e61002171553sff9075dq799c0c68313e9c7c () mail ! gmail ! com
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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@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@lists.digium.com [mailto:asterisk-biz-
> >>bounces@lists.digium.com] On Behalf Of Alex Balashov
> >>Sent: January-06-10 2:29 PM
> >>To: asterisk-biz@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
> >>
> >>_______________________________________________
> >>--Bandwidth and Colocation Provided by http://www.api-digital.com--
> >>
> >>asterisk-biz mailing list
> >>To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
[Attachment #5 (text/html)]
<div>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.</div> <div><br></div><div>Check out here, it's \
free to try.. <a href="http://sipqos.com">http://sipqos.com</a></div><div><br></div><div>Feedback \
always welcome.. good, bad, indifferent.</div><div><br></div><div>- Erik</div>
<br><br><div class="gmail_quote">On Thu, Jan 7, 2010 at 7:05 AM, <span \
dir="ltr"><<a href="mailto:lists@contacttel.com">lists@contacttel.com</a>></span> \
wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px \
#ccc solid;padding-left:1ex;"> Done that as well to test PRI's, but then on some \
small network congestion<br> peeks it would give false positives and reboot boxes \
that were healthy.<br> <br>
Not sure but i guess it's still a good option for that part, as for the<br>
rest, asterisk needs a monitor app built-in, as all other options are<br>
either not powerful enough or too much (requiring a steep learning curve or<br>
config time)<br>
<div class="im"><br>
>>-----Original Message-----<br>
>>From: <a href="mailto:asterisk-biz-bounces@lists.digium.com">asterisk-biz-bounces@lists.digium.com</a> \
[mailto:<a href="mailto:asterisk-biz-">asterisk-biz-</a><br> \
</div><div><div></div><div class="h5">>><a \
href="mailto:bounces@lists.digium.com">bounces@lists.digium.com</a>] On Behalf Of \
Alex Balashov<br> >>Sent: January-06-10 2:29 PM<br>
>>To: <a href="mailto:asterisk-biz@lists.digium.com">asterisk-biz@lists.digium.com</a><br>
>>Subject: Re: [asterisk-biz] Remote SIP monitor<br>
>><br>
>>One thing we've done for a couple customers in the past is write a<br>
>>script that initiates a call (via AMI Originate command) out of a<br>
>>termination provider, which loops back into an origination provider and<br>
>>is received by the same Asterisk instance. Once the call is<br>
>>established, DTMF digits are passed and verified received in both<br>
>>directions.<br>
>><br>
>>If this fails to take place or if the incorrect or incomplete digit<br>
>>sequence is received, an SNMP trap was thrown via System().<br>
>><br>
>>--<br>
>>Alex Balashov - Principal<br>
>>Evariste Systems<br>
>>Web : <a href="http://www.evaristesys.com/" \
target="_blank">http://www.evaristesys.com/</a><br> >>Tel : (+1) (678) \
954-0670<br> >>Direct : (+1) (678) 954-0671<br>
>><br>
>>_______________________________________________<br>
>>--Bandwidth and Colocation Provided by http://www.api-digital.com--<br>
>><br>
>>asterisk-biz mailing list<br>
>>To UNSUBSCRIBE or update options visit:<br>
>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" \
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br> <br>
<br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by http://www.api-digital.com--<br>
<br>
asterisk-biz mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" \
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br> \
</div></div></blockquote></div><br>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
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