I&#39;ve been working at this all day, with no joy.<div><br></div><div>I thought that antivirus software was to blame, but that was a dead end.<span></span><br><div><br></div><div>Does anyone have an example of running the .loc local_stream approach?</div>
<div><br></div><div><br>On Sunday, November 10, 2013, Erik M. Devane - Comms Guy  wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>OK, so I&#39;ve had another chance to work with the settings following your suggestion.<br>
<br>Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards.</div>
<div>SIP trunk from Cisco system, testing using AT&amp;T phone to public phone number. CODEC is PCMU/8000.</div><div><br></div><div>Everything seems perfect when just doing a simple bridge:</div><div>&lt;action application=&quot;bridge&quot; data=&quot;portaudio/endpoint/MAUDIO-${destination_number:-2}&quot;/&gt;<br>

</div><div>I can listen to it for hours, with no issues or delay.</div><div><br></div><div><br></div>Problems start when creating a default conference used, by dialplan:<div><br></div><div><div>&lt;action application=&quot;conference_set_auto_outcall&quot; data=&quot;portaudio/endpoint/MAUDIO-11&quot;/&gt;</div>

<div>&lt;action application=&quot;conference&quot; data=&quot;$1-${domain_name}@${use_profile}++flags{mute}&quot;&quot;/&gt;</div><div><br></div><div>Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.</div>

<div><br></div><div>Setup:</div><div><br></div><div>Conference - default, energy level: 100, waste, all callers set to mute.<br><div><br></div><div>External.xml used for profile, changes to the default:</div><div><div>  </div>

<div>&lt;param name=&quot;rtp-timer-name&quot; value=&quot;none&quot;/&gt;</div><div>&lt;param name=&quot;rtp-autoflush-during-bridge&quot; value=&quot;true&quot;/&gt;</div><div>&lt;param name=&quot;rtp-autoflush&quot; value=&quot;true&quot;/&gt;</div>

</div><div><br></div><div>Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..</div><div><br></div><div>Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.</div>

<div><br></div><div>Any other suggestions would be gladly received. I couldn&#39;t locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.</div>

</div></div><div><br></div><div>Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...</div>

<div><br></div><div><br></div></div><div><br><br><div>On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale <span dir="ltr">&lt;<a>anthony.minessale@gmail.com</a>&gt;</span> wrote:<br>
<blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Are you running the pa and the conference both at a high rate?<div>Some soundcards do bad at slower rates since its emulated.  Its most likely the timing on the soundcard over anything else.</div>

</div><div><div><div>
<br><br><div>On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy <span dir="ltr">&lt;<a>emdevane@gmail.com</a>&gt;</span> wrote:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div dir="ltr">No, I hadn&#39;t - that sounds good. I&#39;m using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.</div><div dir="ltr"><br></div>



<div dir="ltr">Any guidance welcomed!<span></span></div><div><br></div><div>Does anyone have any thoughts on why conferences would be slowing down?</div><div><div>
<div><br>
<div>On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale <span dir="ltr">&lt;<a>anthony.minessale@gmail.com</a>&gt;</span> wrote:<br>

<blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan</div>




<div><br>
<br><div><div><div>On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy <span dir="ltr">&lt;<a>emdevane@gmail.com</a>&gt;</span> wrote:<br>

</div></div><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
I have a very well specified Windows Server connected to sound card input via mod_portaudio.<div><br></div><div>I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.</div>






<div><br></div><div>My assumption is that this is the best way. I am happy to be wrong if there&#39;s a better plan.</div><div><br></div><div>The problem is that delay builds up, seconds every minute on the conference.</div>






<div><br></div><div>I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.</div><div><br></div><div>Any other suggestions I could try? Would CentOS have better timing?</div><div><br></div><div>I don&#39;t need callers to hear each other, but I&#39;m sure that I&#39;m missing something.</div>






<div><br></div><div>Any thoughts?</div><span><font color="#888888"><div><br></div>Erik<br><div><br></div><div><div><br></div></div>
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