<div dir="ltr">Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem.<div>My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.</div>
<div> </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.<div>
<br></div><div>
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen. </div>
<div><br></div><div>However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at </div><div><action application="set" data="conference_enter_sound=silence_stream://10"/><br>
</div><div>but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.</div><div><br></div><div>Is there a way to assign conference params per user (like member flags)?</div>
</div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.<div>
<br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Thanks Anthony!</div><div><br></div>I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.<div>
<br></div><div>Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b</div><div><br></div><div>Would it help if gave you my test ws server and sip address so you could reproduce this? </div>
<div><br></div><div>Is it possible that the problem is with my AWS instance or CentOS 5.9?</div><div><br></div><div><br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>I set up your test ext here:</div><div><a href="https://webrtc.freeswitch.org/sipjs/index.html#8765" target="_blank">https://webrtc.freeswitch.org/sipjs/index.html#8765</a><br>
</div><div><br></div><div>And it seems to work.</div>
<div><br></div>Try this in vars.xml<div><br></div><div>
<p><span><</span><span>X-PRE-PROCESS</span><span> </span><span>cmd</span><span>=</span>"set"<span> </span><span>data</span><span>=</span>"suppress_cng=true"<span>/> </span></p>
<p><span>I don't think the silence suppression works very well in the browser.</span></p><p><span><br></span></p><p><span><br></span></p></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote"><div><div>On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
<div dir="ltr">In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.</div>
<div><div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi guys,<div><br></div><div>I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:</div><div><br></div><div>
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.</div>
<div><br></div><div>2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.</div><div><br></div><div>This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.</div>
<div><br></div><div>I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.</div>
<div><br></div><div>Has anyone run into this? Any advice?</div><div><br></div><div>Here is the relevant excerpt from my dialplan.</div><div><br></div><div><condition field="destination_number" expression="^conf-(\S+)$"><br>
</div><div> <action application="set" data="answer_delay=3000"/><br></div><div><div> <span style="white-space:pre-wrap">        </span><action application="answer"/></div>
<div> <action application="playback" data="tone_stream://%(1000,0,600)" /></div><div> <action application="conference" data="$1@default"/></div><div></condition></div>
</div><div><br></div></div>
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