[asterisk-biz] Remote SIP monitor

[lists at contacttel.com]

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
>>
>>_______________________________________________
>>--Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
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