Difference between revisions of "Telephone system:shoutcast streams"

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m (added category)
(updated stream conference logic (they are not numbered anymore))
 
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<onlyinclude>The most efficient way to have shoutcast streams as <abbr title="music on hold">MOH</abbr> is to set up a muted conference. This way, the fist connection will be made when the first listener joins, and no extra streams will be opened.
 +
</onlyinclude>
 +
 
=== Synopsis ===
 
=== Synopsis ===
  
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The setup mentioned below will start one stream for the first caller, and will close it after the last caller leaves.
 
The setup mentioned below will start one stream for the first caller, and will close it after the last caller leaves.
 +
{{StreamLinkTable
 +
|stages=Saal Adams (13x1);s1,Saal Borg (13x2);s2,Saal Clarke (13x3);s3,Saal Dijkstra (13x4);s4,Saal Eliza (13x5);s5,Chaos West Bühne (13x6);s150,WikiPakaWG Esszimmer (13x7);s89,Open Infrastructure Orbit (13x8);oio
 +
|formats=Audio;.mp3|types=Native;|typeCount=1
 +
|languages=;_native|languageCount=1
 +
|}}
  
−
=== implementation ===
+
== implementation ==
 +
 
 +
=== totally muted single stream ===
 
==== dialplan ====
 
==== dialplan ====
−
This sample is the 1301 to 1303, used for dialing in one of three rooms when a conference is happening.
+
This sample is the 1300 to 1309, used for dialing in one of nine rooms when a conference is happening.
 
When there currently is no event, the <abbr title="special information tone">SIT</abbr> will be played
 
When there currently is no event, the <abbr title="special information tone">SIT</abbr> will be played
  
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<pre>
 
<pre>
−
     <extension name="event">
+
     <!-- These are moderated muted conferences, which will give you the single MOH stream without moderator
−
      <condition field="destination_number" expression="^130([1-3])$">
 
−
        <action application="answer"/>
 
−
        <!-- This is a moderated muted conference, which will give you the single MOH stream without moderator
 
 
             make sure moh is not set in the conference params, so we can set it here
 
             make sure moh is not set in the conference params, so we can set it here
 
             also, don't set the enter and leave (global) sounds, since it will reconnect the stream -->
 
             also, don't set the enter and leave (global) sounds, since it will reconnect the stream -->
  
−
         <!--action application="set" data="conference_moh_sound=shout://someserver/$1.mp3"/-->
+
    <!-- by default, play SIT to indicate the stream is not available -->
−
         <action application="set" data="conference_moh_sound=tone_stream://$${sit}"/>
+
    <extension name="unavailable event" continue="true">
 +
      <condition field="destination_number" expression="^130([0-9])$">
 +
         <action application="set" data="event[0]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[1]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[2]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[3]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[4]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[5]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[6]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[7]=tone_stream://$${nl-sit}" inline="true"/>
 +
        <action application="set" data="event[8]=tone_stream://$${nl-sit}" inline="true"/>
 +
         <action application="set" data="event[9]=tone_stream://$${nl-sit}" inline="true"/>
 +
      </condition>
 +
    </extension>
  
 +
    <!-- classic example of numbered rooms -->
 +
    <!--extension name="event stream numbered">
 +
      <condition field="destination_number" expression="^130([1-4])$">
 +
        <action application="set" data="event[${1}]=shout://live.self.c3voc.de/S${1}_native.mp3" inline="true"/>
 +
      </condition>
 +
    </extension-->
 +
 +
    <!-- actual conferences: just comment out the streams that are not available -->
 +
    <extension name="event stream">
 +
      <condition field="destination_number" expression="^130([0-9])$">
 +
        <action application="answer"/>
 +
 +
        <!--vv list of conferences here vv-->
 +
        <action application="set" data="event[0]=shout://playerservices.streamtheworld.com/api/livestream-redirect/KINK.mp3" inline="true"
 +
        <action application="set" data="event[1]=shout://live.self.c3voc.de/c3lounge_native.mp3" inline="true"/>
 +
        <action application="set" data="event[2]=shout://live.self.c3voc.de/abchillgleis_native.mp3" inline="true"/>
 +
        <!--^^ list of conferences here ^^-->
 +
 +
        <!-- conference_moh_sound is set earlier in this dialplan file -->
 +
        <action application="set" data="conference_moh_sound=${event[${1}]}" inline="true"/>
 
         <action application="conference" data="130$1@stream+flags{mute}"/>
 
         <action application="conference" data="130$1@stream+flags{mute}"/>
 
       </condition>
 
       </condition>
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</pre>
 
</pre>
  
−
[[Category:Telephony]]
+
=== party line listening to the muted stream ===
 +
==== dialplan ====
 +
This sample is the 1391 to 1399, used for dialing in one of nine rooms when a conference is happening.
 +
When there currently is no event, the <abbr title="special information tone">SIT</abbr> will be played
 +
 
 +
When the first user calls in, it will automatically add the streaming MOH conference, so that they can chat while the stream is playing on the background.  It will stop the conference when the last (real) person hangs up.
 +
 
 +
<pre>
 +
    <extension name="event partyline">
 +
      <condition field="destination_number" expression="^139([0-9])$">
 +
        <action application="answer"/>
 +
        <!-- This is an unmoderated conference, which will give you the single MOH stream
 +
            don't set moh in the conference params -->
 +
 
 +
        <action application="conference_set_auto_outcall" data="loopback/130$1"/>
 +
 
 +
        <action application="conference" data="139$1@partystream+flags{mintwo}"/>
 +
      </condition>
 +
    </extension>
 +
</pre>
 +
 
 +
==== conference.conf.xml profile ====
 +
For this to work, a separate conference profile needs to be used:
 +
It doesn't have a MOH set, since it adds the MOH stream conference
 +
Also, it doesn't have announcements sound set, so the stream doesn't reconnect after a user enters or leaves.
 +
 
 +
<pre>
 +
    <profile name="partystream">
 +
      <param name="domain" value="$${domain}"/>
 +
      <param name="rate" value="32000"/>
 +
      <param name="interval" value="20"/>
 +
      <param name="energy-level" value="300"/>
 +
      <param name="muted-sound" value="conference/conf-muted.wav"/>
 +
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
 +
      <param name="pin-sound" value="conference/conf-pin.wav"/>
 +
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
 +
      <param name="caller-id-name" value="$${outbound_caller_name}"/>
 +
      <param name="caller-id-number" value="$${outbound_caller_id}"/>
 +
     
 +
      <param name="comfort-noise" value="false"/>
 +
    </profile>
 +
</pre>
 +
 
 +
[[Category:Telephony]][[Category:Telephone snippet]][[Category:FreeSWITCH]]

Latest revision as of 19:59, 27 December 2021

The most efficient way to have shoutcast streams as MOH is to set up a muted conference. This way, the fist connection will be made when the first listener joins, and no extra streams will be opened.


Synopsis

ACKspace has several shoutcast streams you can dial in to. To see the list of streams, see the dialplan

Normally, the streams are put in a local stream.loc file, but this will cause FS to open all streams, even when there are no listeners. Since we have so many, this is not an option.

Also, when you set a MOH per user, this will open the same stream simultaneously for each caller.

The setup mentioned below will start one stream for the first caller, and will close it after the last caller leaves.


Audio
Native
Saal Adams (13x1) link
Saal Borg (13x2) link
Saal Clarke (13x3) link
Saal Dijkstra (13x4) link
Saal Eliza (13x5) link
Chaos West Bühne (13x6) link
WikiPakaWG Esszimmer (13x7) link
Open Infrastructure Orbit (13x8) link


implementation

totally muted single stream

dialplan

This sample is the 1300 to 1309, used for dialing in one of nine rooms when a conference is happening. When there currently is no event, the SIT will be played

The users are muted, but also will there never be a moderator, which causes the other callers to listen to MOH indefinitely. 

    <!-- These are moderated muted conferences, which will give you the single MOH stream without moderator
             make sure moh is not set in the conference params, so we can set it here
             also, don't set the enter and leave (global) sounds, since it will reconnect the stream -->

    <!-- by default, play SIT to indicate the stream is not available -->
    <extension name="unavailable event" continue="true">
      <condition field="destination_number" expression="^130([0-9])$">
        <action application="set" data="event[0]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[1]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[2]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[3]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[4]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[5]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[6]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[7]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[8]=tone_stream://$${nl-sit}" inline="true"/>
        <action application="set" data="event[9]=tone_stream://$${nl-sit}" inline="true"/>
      </condition>
    </extension>

    <!-- classic example of numbered rooms -->
    <!--extension name="event stream numbered">
      <condition field="destination_number" expression="^130([1-4])$">
        <action application="set" data="event[${1}]=shout://live.self.c3voc.de/S${1}_native.mp3" inline="true"/>
      </condition>
    </extension-->

    <!-- actual conferences: just comment out the streams that are not available -->
    <extension name="event stream">
      <condition field="destination_number" expression="^130([0-9])$">
        <action application="answer"/>

        <!--vv list of conferences here vv-->
        <action application="set" data="event[0]=shout://playerservices.streamtheworld.com/api/livestream-redirect/KINK.mp3" inline="true"
        <action application="set" data="event[1]=shout://live.self.c3voc.de/c3lounge_native.mp3" inline="true"/>
        <action application="set" data="event[2]=shout://live.self.c3voc.de/abchillgleis_native.mp3" inline="true"/>
        <!--^^ list of conferences here ^^-->

        <!-- conference_moh_sound is set earlier in this dialplan file -->
        <action application="set" data="conference_moh_sound=${event[${1}]}" inline="true"/>
        <action application="conference" data="130$1@stream+flags{mute}"/>
      </condition>
    </extension>


conference.conf.xml profile

For this to work, a separate conference profile needs to be used: It doesn't have a MOH set, so it can be overridden at the dial plan. Also, it doesn't have announcements sound set, so the stream doesn't reconnect after a user enters or leaves. 

    <profile name="stream">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="32000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="300"/>
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      <param name="caller-id-name" value="$${outbound_caller_name}"/>
      <param name="caller-id-number" value="$${outbound_caller_id}"/>
      
      <param name="comfort-noise" value="false"/>
      <param name="conference-flags" value="wait-mod"/>
      <param name="caller-controls" value="none"/>
    </profile>

party line listening to the muted stream

dialplan

This sample is the 1391 to 1399, used for dialing in one of nine rooms when a conference is happening. When there currently is no event, the SIT will be played

When the first user calls in, it will automatically add the streaming MOH conference, so that they can chat while the stream is playing on the background. It will stop the conference when the last (real) person hangs up. 

    <extension name="event partyline">
      <condition field="destination_number" expression="^139([0-9])$">
        <action application="answer"/>
        <!-- This is an unmoderated conference, which will give you the single MOH stream
             don't set moh in the conference params -->

        <action application="conference_set_auto_outcall" data="loopback/130$1"/>

        <action application="conference" data="139$1@partystream+flags{mintwo}"/>
      </condition>
    </extension>

conference.conf.xml profile

For this to work, a separate conference profile needs to be used: It doesn't have a MOH set, since it adds the MOH stream conference Also, it doesn't have announcements sound set, so the stream doesn't reconnect after a user enters or leaves. 

    <profile name="partystream">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="32000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="300"/>
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      <param name="caller-id-name" value="$${outbound_caller_name}"/>
      <param name="caller-id-number" value="$${outbound_caller_id}"/>
      
      <param name="comfort-noise" value="false"/>
    </profile>
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