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Asked: 2009-12-12 19:39:18 +0800 CST

Ways to monitor SIP termination on an asterisk server

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I have a nagios setup which ensures that SIP is responsive on my Asterisk server, that's straight forward.

My question is, what kind of possibilities are there that the Asterisk server can actually terminate properly with the termination provider? As in produce a test call and ensure that something on the other end picks up?

I realize this is a multi-part and very difficult problem. Without expensive equipment, it's almost impossible to tell what medium is on the other end of termination point.

Just curious what you SIP/Asterisk guys are doing to monitor for this to know when there is an outage besides your users calling you to tell you that's the case?

monitoring

2 Answers

  1. Best Answer

    I have come across the same problem and it's not just limited to asterisk. In the end we came up with something that worked well for us. We called it a SIP Loopback.

    Basically we signed up for an ALT sip provider (flowroute.com) and setup a script that calls out via primary SIP provider to the phone number setup with our ALT provider three times an hour. The incoming phone call would check CALLERID and then if was from primary phone number post to nagios a succes message. If no succes messages were posted to nagios at least once ever hour it would alarm. You would need to write the scripts yourself as I no longer have access to them myself but it should not be hard.

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  2. You can use online service from www.sipnms.com to send sip messages and if there is a difference in the message received it will alarm you using email or snmp. you can define message scripts with sip options or invites.

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