I went with sipp for that exact same purpose. I had to script a bit around it to integrate with my zabbix monitoring but now I can get alerts on failed calls. On Wed, Jun 9, 2010 at 5:48 PM, John Todd <jtodd at loligo.com> wrote:
Perhaps "recqual" is what you're looking for. ?End-to-end testing with tones, comparing the recorded tone with the transmitted tone. ?Works with Asterisk. ? Implies you can control both "end" of the call, since otherwise it is extremely difficult to conclude that a call is "completed" since many errors are not obvious if all you're looking for is a media path starting up.
http://www.voip-info.org/wiki/view/Recqual
JT
On Jun 9, 2010, at 1:27 PM, Graham Freeman wrote:
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
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops