[Freeswitch-users] Help needed with audio test setup

Cal Leeming [Simplicity Media Ltd] cal.leeming at simplicitymedialtd.co.uk
Mon Sep 30 23:57:09 MSD 2013


Thanks very much for sharing, glad you were able to get it sorted and
hopefully this helps others in the future!

Cal


On Mon, Sep 30, 2013 at 5:37 PM, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

>  Hi,
>
> It's done and works for me. Was a fun thing to do indeed! Dialplan portion
> of it is documented here in the very bottom:
>
> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect#Samples
>
> The controlling script portion is in bash and along these lines
> -query if old test calls are already up with "show calls". kill any old
> test calls with "uuid_kill"
> -launch N new test calls using "originate"
> -wait 3 minutes, check with "show calls" if all test calls ended
> succesfully. If so, start the test routine again.
> -if not, stuck test call means attention required. Get call stats with
> "uuid_dump", run tcpdump on the rtp port and IPs of that call, email result.
>
> Hope this helps!
>
> -Victor
>
>  On 13-09-30 04:13 AM, Cal Leeming [Simplicity Media Ltd] wrote:
>
> This would be quite easy to throw together, and would be a fun task for
> you to take on :)
>
>  Once you've finished it, you should consider sharing it with the
> community as I've seen this question come up a few times, it would be nice
> to give others this functionality out of the box.
>
>  Let us know how you get on
>
>  Cal
>
>
> On Fri, Sep 27, 2013 at 5:57 PM, Victor Chukalovskiy <
> victor.chukalovskiy at gmail.com> wrote:
>
>> Hello Community,
>>
>> The problem is that a 3-rd party SIP hardware causes audio isssues in
>> less then 1% of the cases. Need to do 100's of manual test calls to
>> identify it and bring to the vendor.
>>
>> Is there a quick recepie to make the following FS test setup? Should be
>> as simple as possible. If someone has something ready-made to offer,
>> please contact me off-list.
>>
>> Here is the idea:
>>
>> -FS generates test SIP call and keeps it up for N minutes.
>> -One side initates the call and sends test tone at X Hz
>> -another side (also FS) answers the call and sends test tone at Y Hz
>> -both sides listen to the tone coming from the other side.
>> -If tone in either direction degrades into siginificant noise or
>> silence, FS leaves the call up indifentely long AND sends email
>> notification "got the sucker!"
>> -If in N minutes tone in both directions did not degrade, FS hangs up
>> the test call and initates a new one. Infinite testing cycle until we
>> turn it off.
>>
>> Thank you,
>> Victor
>>
>> _________________________________________________________________________
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>
>
>
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