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: [email protected] [mailto:asterisk-biz- >>[email protected]] On Behalf Of Alex Balashov >>Sent: January-06-10 2:29 PM >>To: [email protected] >>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
