One of the most bizzare things about sipp is its placement of XML configurations into string constants in the code. On Jun 11, 2010, at 12:27 AM, Antoine Reversat <a.reversat at gmail.com> wrote:
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
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