[VoiceOps] Automated call completion testing?
John Todd
jtodd at loligo.com
Wed Jun 9 17:48:54 EDT 2010
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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