<div dir="ltr"><div><div><div>is it running on a virtual machine?<br><br></div>I found a strange effect that I could only reproduce in a VM, and never on physical hardware:<br><a href="https://freeswitch.org/jira/browse/FS-7805">https://freeswitch.org/jira/browse/FS-7805</a><br></div>under certain load, an originate command triggers a continuous distortion in another, running and unrelated, channel.<br><br></div>It seems to be triggered by insufficient CPU resource at the moment of the origination.<br><br><br><br><br><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <span dir="ltr"><<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@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've found that it occurs after any bridge, its not specific to the originate command. <br></div><div><br></div><div>Would you require a wireshark trace or the output of the freeswitch console?</div><div><br></div><div>The scenario below reproduces the issue ...<br></div><div><br></div><div>Test extension</div><div><br></div><div><div><extension name="Inbound_Routing"></div><div> <condition field="destination_number" expression="^2003$"></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="dialed_extension=$1"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="transfer_ringback=$${uk-ring}"/></div><div><span style="white-space:pre-wrap"> </span><action application="set" data="RECORD_TITLE=Title goes here"/></div><div><span style="white-space:pre-wrap"> </span><action application="set" data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="send_silence_when_idle=false"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="bridge_generate_comfort_noise=false"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="rtp_enable_vad_in=false"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="rtp_enable_vad_out=false"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="fire_talk_events=true"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="fire_not_talk_events=true"/></div><div><span style="white-space:pre-wrap"> </span><action application="export" data="recording_id=${uuid}"/></div><div><span style="white-space:pre-wrap"> </span><action application="set" data="media_bug_answer_req=true"/></div><div><span style="white-space:pre-wrap"> </span><action application="answer" /></div><div><span style="white-space:pre-wrap"> </span><action application="playback" data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/></div><div> </condition></div><div></extension></div></div><div><div><br></div><div>I originate a call from a polycom handset using g722 to the extension above.</div><div>I originate a call using the verto client to the extension above.</div><div><br></div><div>I get the uuids of the channels using show channels and use uuid_bridge [uuid1] [uuid2] to merge the channels.</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 1, 2015 at 11:30 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Send FreeSWITCH-users mailing list submissions to<br>
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than "Re: Contents of FreeSWITCH-users digest..."<br>
<br>Today's Topics:<br>
<br>
1. How send in To header anonymous and in P-Asserted-Identity<br>
caller-id-number (Alex Polischuk)<br>
2. Loop play while wait DTMF digit (Dmitry Mordovin)<br>
3. Broken silence with webrtc (Gary Foreman)<br>
4. Re: Broken silence with webrtc (Brian West)<br>
5. SIP profile not loading (Chris Young)<br>
6. Playing with conditions (Dmitry Mordovin)<br>
7. Re: Playing with conditions (Avi Marcus)<br>
<br><br>---------- Forwarded message ----------<br>From: Alex Polischuk <<a href="mailto:alxpol@gmail.com" target="_blank">alxpol@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Mon, 31 Aug 2015 17:02:25 +0300<br>Subject: [Freeswitch-users] How send in To header anonymous and in P-Asserted-Identity caller-id-number<br><div dir="ltr">Hi all,<div><br></div><div>How I can define different users and domains in To and <span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;line-height:20px">P-Asserted-Identity headers?</span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;line-height:20px"><br></span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;line-height:20px">Thanks,</span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;line-height:20px">Alex</span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;line-height:20px"><br></span></div></div>
<br><br>---------- Forwarded message ----------<br>From: Dmitry Mordovin <<a href="mailto:d.mordovin@dwide.com" target="_blank">d.mordovin@dwide.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Mon, 31 Aug 2015 18:24:08 +0400<br>Subject: [Freeswitch-users] Loop play while wait DTMF digit<br>
<div bgcolor="#FFFFFF" text="#000000">
Hello<br>
<br>
This example play conf-pin.wav and wait DTMF.<br>
<br>
<pre style="font-family:monospace,Courier;padding:1em;border:1px dashed rgb(47,111,171);color:rgb(0,0,0);line-height:1.3em;font-size:12.8000001907349px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing:0px;background-color:rgb(249,249,249)"><extension name="play_and_get_digits example">
<condition field="destination_number" expression="^(1888)$">
<action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/>
<action application="log" data="CRIT ${foobar}"/>
</condition>
</extension></pre>
<br>
Is it possible to play WAV file in infinity loop and wait user DTMF?<br>
<br>
And how can I check DTMF input after user entered DTMF?<br>
<br>
In dialpeer, like.<br>
If ${DTMF} = 1 then bridge to XXX<br>
If ${DTMF} = 2 then play file and finish session<br>
<br>
<br>
<br>
Thank you.<br>
Dmitry<br>
<br>
</div><span class="">
<br><br>---------- Forwarded message ----------<br>From: Gary Foreman <<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@gmail.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Cc: <br>Date: Mon, 31 Aug 2015 20:36:59 +0100<br>Subject: [Freeswitch-users] Broken silence with webrtc<br><div dir="ltr">Hi all, <div><br></div><div>Hoping someone can point me in the right direction because after several hours I'm out of ideas.</div><div><br><div>I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command.</div><div><br></div><div>Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream.</div><div><br></div><div>Thanks in advance!</div><div><br></div><div><br><div><br></div><div><br></div></div></div></div>
<br><br></span><div><div class="h5">---------- Forwarded message ----------<br>From: Brian West <<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Mon, 31 Aug 2015 14:40:17 -0500<br>Subject: Re: [Freeswitch-users] Broken silence with webrtc<br><div dir="ltr">Please provide logs and samples of how you originate this, sounds like Voice Activity Detection possibly.</div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <span dir="ltr"><<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Hi all, <div><br></div><div>Hoping someone can point me in the right direction because after several hours I'm out of ideas.</div><div><br><div>I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command.</div><div><br></div><div>Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream.</div><div><br></div><div>Thanks in advance!</div><div><br></div><div><br><div><br></div><div><br></div></div></div></div>
<br>_________________________________________________________________________<br>
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<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font face="courier new, monospace" size="1"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br></font></p><p><font face="monospace, monospace" size="2"><b><i>Twitter: @FreeSWITCH , @briankwest</i></b><br><a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br><a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p><p><font face="monospace, monospace">Got Bugs? Report them <a href="https://freeswitch.org/jira" target="_blank">here</a>! | Reddit: <a href="https://www.reddit.com/r/freeswitch" target="_blank">/r/freeswitch</a></font></p>
<p><font face="monospace, monospace" size="2"><b>T:</b><a href="tel:%2B19184209001" value="+19184209001" target="_blank">+19184209001</a> | <b>F:</b><a href="tel:%2B19184209002" value="+19184209002" target="_blank">+19184209002</a> | <b>M:</b>+1918424WEST (9378)<br><b>iNUM:</b>+883 5100 1420 9001 | <b>ISN:</b>410*543 | <b>Skype:</b>briankwest</font></p></div></div></div></div></div></div></div></div></div></div>
</div>
<br><br></div></div>---------- Forwarded message ----------<br>From: Chris Young <<a href="mailto:Chris.Young@enghouse.com" target="_blank">Chris.Young@enghouse.com</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Tue, 1 Sep 2015 07:31:00 +0000<br>Subject: [Freeswitch-users] SIP profile not loading<br>
<div link="blue" vlink="purple" lang="EN-GB">
<div>
<p class="MsoNormal">Hi all,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named
(for example):<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"> dummy.xml<u></u><u></u></p>
<p class="MsoNormal"> external.xml<u></u><u></u></p>
<p class="MsoNormal"> internal.xml<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address
is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly
so it appears to be valid.<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Has anybody else seen this kind of behaviour or know what could be causing it?<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Many thanks,<u></u><u></u></p>
<p class="MsoNormal">Chris<span><u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
</div>
<br><br>---------- Forwarded message ----------<br>From: Dmitry Mordovin <<a href="mailto:d.mordovin@dwide.com" target="_blank">d.mordovin@dwide.com</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Tue, 01 Sep 2015 12:56:08 +0400<br>Subject: [Freeswitch-users] Playing with conditions<br>Hello<br>
<br>
<section name="dialplan" description="Local call"><br>
<context name="public"><br>
<extension name="local_1"><br>
<br>
<condition field="destination_number" expression="^(\d{4})$"><br>
<action application="answer"/><br>
<br>
<action application="play_and_get_digits" data="1 1 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/><br>
<action application="log" data="CRIT ${foobar}"/><br>
<br>
<condition field="${foobar}" expression="1$"><br>
<action application="log" data="CRIT --------------1------------------"/><br>
<anti-action application="log" data="CRIT --------------not 1------------------"/><br>
</condition><br>
<br>
<action application="log" data="CRIT ---- last ${foobar}"/><br>
</condition><br>
<br>
</extension><br>
</context><br>
</section><br>
<br>
Why inner condition not works?<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Avi Marcus <<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Tue, 1 Sep 2015 10:30:00 +0000<br>Subject: Re: [Freeswitch-users] Playing with conditions<br><div dir="ltr">Short answer: Each extension only has 1 set of conditions.<div>The condition evaluating foobar is run <i>before</i> it gets set.</div><div>After the <span style="font-size:12.8000001907349px">play_and_get_</span><span style="font-size:12.8000001907349px">digits you should transfer/execute_extension to a new extension that will evaluate ${foobar}</span></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">-Avi Marcus<br>BestFone</div></div></div>
<br><div class="gmail_quote">On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <span dir="ltr"><<a href="mailto:d.mordovin@dwide.com" target="_blank">d.mordovin@dwide.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Hello<br>
<br>
<section name="dialplan" description="Local call"><br>
<context name="public"><br>
<extension name="local_1"><br>
<br>
<condition field="destination_number" expression="^(\d{4})$"><br>
<action application="answer"/><br>
<br>
<action application="play_and_get_digits" data="1 1 3 7000 #<br>
$${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav<br>
/invalid.wav foobar \d+"/><br>
<action application="log" data="CRIT ${foobar}"/><br>
<br>
<condition field="${foobar}" expression="1$"><br>
<action application="log" data="CRIT<br>
--------------1------------------"/><br>
<anti-action application="log" data="CRIT --------------not<br>
1------------------"/><br>
</condition><br>
<br>
<action application="log" data="CRIT ---- last ${foobar}"/><br>
</condition><br>
<br>
</extension><br>
</context><br>
</section><br>
<br>
Why inner condition not works?<span class=""><br>
<br>
_________________________________________________________________________<br>
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</span></blockquote></div><br></div></div>
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<br></span></blockquote></div><br></div></div></div></div></div>
<br>_________________________________________________________________________<br>
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