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<div class="moz-cite-prefix">Hmm, this may work. Does it generate a
channel variable once complete?<br>
<br>
Also, I see hits a counted in frames. What are the units? 20 ms
frame or something else?<br>
<br>
On 13-09-27 02:57 PM, Michael Jerris wrote:<br>
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<a moz-do-not-send="true"
href="https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence">https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence</a>
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<div>On Sep 27, 2013, at 2:34 PM, Victor Chukalovskiy <<a
moz-do-not-send="true"
href="mailto:victor.chukalovskiy@gmail.com">victor.chukalovskiy@gmail.com</a>>
wrote:</div>
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<div class="moz-cite-prefix">Clarification:<br>
<br>
No need to measure RTP quality. Just need to detect when
a test tone changes into white noise or dead air.<br>
<br>
I see mod_spandsp can fire events on tone detection. Is
there a way to do inverse? That is, fire event on tone
loss?<br>
<br>
On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:<br>
</div>
<blockquote
cite="mid:1380302556.1196.YahooMailNeo@web126201.mail.ne1.yahoo.com"
type="cite">
<div style="background-color: rgb(255, 255, 255);
font-family: arial, helvetica, sans-serif; font-size:
10pt; ">should be quite easy to do, probably a day of
scripting.<br>
The only issue, how to measure the RTP quality in real
time.<br>
<br>
<br>
<div><span><br>
</span></div>
<div><br>
</div>
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font-size: 10pt;">
<div style="font-family: times new roman, new york,
times, serif; font-size: 12pt;">
<div dir="ltr">
<hr size="1"> <font face="Arial" size="2"> <b><span
style="font-weight:bold;">From:</span></b>
Victor Chukalovskiy <a moz-do-not-send="true"
class="moz-txt-link-rfc2396E"
href="mailto:victor.chukalovskiy@gmail.com"><victor.chukalovskiy@gmail.com></a><br>
<b><span style="font-weight: bold;">To:</span></b>
FreeSWITCH Users Help <a
moz-do-not-send="true"
class="moz-txt-link-rfc2396E"
href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a>
<br>
<b><span style="font-weight: bold;">Sent:</span></b>
Friday, September 27, 2013 6:57 PM<br>
<b><span style="font-weight: bold;">Subject:</span></b>
[Freeswitch-users] Help needed with audio test
setup<br>
</font> </div>
<div class="y_msg_container"><br>
Hello Community,<br>
<br>
The problem is that a 3-rd party SIP hardware
causes audio isssues in <br>
less then 1% of the cases. Need to do 100's of
manual test calls to <br>
identify it and bring to the vendor.<br>
<br>
Is there a quick recepie to make the following
FS test setup? Should be <br>
as simple as possible. If someone has something
ready-made to offer, <br>
please contact me off-list.<br>
<br>
Here is the idea:<br>
<br>
-FS generates test SIP call and keeps it up for
N minutes.<br>
-One side initates the call and sends test tone
at X Hz<br>
-another side (also FS) answers the call and
sends test tone at Y Hz<br>
-both sides listen to the tone coming from the
other side.<br>
-If tone in either direction degrades into
siginificant noise or <br>
silence, FS leaves the call up indifentely long
AND sends email <br>
notification "got the sucker!"<br>
-If in N minutes tone in both directions did not
degrade, FS hangs up <br>
the test call and initates a new one. Infinite
testing cycle until we <br>
turn it off.<br>
<br>
Thank you,<br>
Victor<br>
<br>
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Official FreeSWITCH Sites
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