Telephone system:shoutcast streams

From Hackerspace ACKspace
Jump to: navigation, search

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.

implementation

totally muted single stream

dialplan

This sample is the 1301 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.

    <extension name="event">
      <condition field="destination_number" expression="^130([1-9])$">
        <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
             also, don't set the enter and leave (global) sounds, since it will reconnect the stream -->

        <action application="set" data="conference_moh_sound=tone_stream://$${sit}"/>
        <!--action application="set" data="conference_moh_sound=shout://someserver/$1.mp3"/-->

        <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([1-9])$">
        <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>