<div dir="ltr"><div><a href="https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc">https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc</a><br></div><div><br></div><div>Its just setting up mod_local_stream to point at a dir with a file with a .loc extension that has the string of the pa url in it. </div>
<div><br></div>You should probably not change any of those params away from default. I would recommend putting those back to where they belong.<div><br></div><div>If anything it could worsen your problem. What about the interval? DId you try increasing that?</div>
<div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy <span dir="ltr"><<a href="mailto:emdevane@gmail.com" target="_blank">emdevane@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I'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><div class="h5">
<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'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&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><action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/><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><action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/></div>
<div><action application="conference" data="$1-${domain_name}@${use_profile}++flags{mute}""/></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><param name="rtp-timer-name" value="none"/></div><div><param name="rtp-autoflush-during-bridge" value="true"/></div><div><param name="rtp-autoflush" value="true"/></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'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"><<a>anthony.minessale@gmail.com</a>></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"><<a>emdevane@gmail.com</a>></span> wrote:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">No, I hadn't - that sounds good. I'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"><<a>anthony.minessale@gmail.com</a>></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"><<a>emdevane@gmail.com</a>></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'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't need callers to hear each other, but I'm sure that I'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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