<div dir="ltr">I am doing a similar setup on a Windows Server using conference with some C# code.In my system I have a few extension each one is playing music from a different web stream. The way I set it up is that when a call comes in I check if the conference related to the dialed extension is already setup ("conference <extension> list" doesn't return "not found"), if not I create a new thread and execute "conference <extension> play <url> async" and then conference in the caller.<div>
If the conference is already in session I just add the caller to the conference.</div><div><br></div><div>The reason I do it on a separate thread is that conference play only works once the conference is already created, but the api call to add the caller to the conference is a blocking call and will only return when the user hangs up or is kicked from the conference.</div>
<div><br></div><div>I also had some issues with the stream falling behind, also the stream would sometimes go silent, so I resolved this by setting a time for every 5 minutes which executes "conference <extension> stop" and then "conf <extension> play <url>". This happens so quickly that the listeners barely notice that the stream has been restarted. </div>
<div><br></div><div>You might be able to implement a similar setup with other languages. I went through a few iterations until I got to this setup and it now appears to be working great. </div><div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Fri, Nov 22, 2013 at 7:55 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">
Increasing the conference internal to 10 to 20 to <span></span>200 had no detectable effect.<div><br></div><div>In order to improve timing, I rebuilt the server as Centos 6.4, building from the tarball (git had some proxy-related issues).</div>
<div><br></div><div>[New issue on Linux]</div><div>I haven't been able to get mod_portaudio to give me more than two mono inputs per M-Audio Delta 1010 card (though alsamixer and pa_devs seem perfectly happy).<br><div>
<br></div><div><br></div><div>[Back to the original issue]</div><div>I reverted all RTP timer, flush, and other changes, back very close to the default configuration.</div><div><br></div><div>Still I have a delay, still increases seconds per minute.</div>
<div><br></div><div>I explored moh but couldn't figure out how to have multiple playlists. If this is possible, Id love to know how!</div><div><br></div><div>My best solution so far has been to have the first call bridge portaudio and subsequent calls eavesdrop on that first bridged call. Latency is almost as good as a single bridged call, and I have tested six eavesdroppers with no ill effects.</div>
<div><br></div><div>Right now I'm going to:</div><div><br></div><div>1) Work out how to mute external caller's audio from the eavesdrop.</div><div><br></div><div>2) Learn lua enough to roll an eavesdropper onto the bridge when the first caller hangs up.</div>
<div><br></div><div>It's not a satisfactory solution, but I'm out of ideas for the conference lag unless anyone else can help, and I don't know how to do multiple MOH instances.</div><div><br></div><div>Any help would be most appreciated - either on conference lag or moh or eavesdrop mute.</div>
<div><br></div><div>Erik</div><div><br>On Tuesday, November 12, 2013, Anthony Minessale wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>
<a href="https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc" target="_blank">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>On Mon, Nov 11, 2013 at 11:00 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">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>
<div><br></div><div><br>On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:<br><blockquote 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@g</a></span></div></div></div></div></blockquote></div></div></div></div></blockquote></div></div></div></div>
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