Ditto on sipsak and rolling your own. An additional tool comes from way back in the Age of the Mullet. The "internet edge" was mostly dialup modems back then. If you're willing to slap an ATA on one end, you can find a bunch of modem (war dialing, dialback, shared pool, etc) scripts out there. David On Wed, Jun 9, 2010 at 1:55 PM, Scott Berkman <scott at sberkman.net> wrote:
I'd probably start with SIPsak (http://sipsak.org/) but it would be a tool in a script you wrote probably, not the end all be all. ?I'm not sure I've seen an open source solution that matches these requirements. ?In the paid world look at Spirent's Abacus.
? ? ? ?-Scott
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Graham Freeman Sent: Wednesday, June 09, 2010 4:28 PM To: voiceops at voiceops.org Subject: [VoiceOps] Automated call completion testing?
Hi, folks,
I'm on the hunt for an automated call completion testing solution.
Criteria:
Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter.
Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold
Must be able to test legacy phone numbers (e.g. +14154622991)
Must be able to test via configurable routes
Should be open-source
Should be compatible with Asterisk
Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense
Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991).
Any suggestions?
thanks,
Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
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