[VoiceOps] Automated call completion testing?

[Antoine Reversat]

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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>
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