Any Success story of streaming Asterisk to an icecast2 serv?

The goal is to stream some incoming phone calls. Is there any success story?



I did have this working once upon a time following these directions: … k+cmd+Ices

But do not actively use it right now. What problems are you having?

Thanks MuppetMaster! Is there an example of asterisk/contrib/asterisk-ices.xml?

I have not tested it of late, so while I am posting this example I do not warranty the fact that it will actually work:

[quote=“asterisk-ices.xml”]<?xml version="1.0"?>

<!-- run in background  -->
<!-- where logs go. -->
<!-- 1=error, 2=warn, 3=infoa ,4=debug -->
<!-- logfile is ignored if this is set to 1 -->

<!-- optional filename to write process id to -->
<!-- <pidfile>/home/ices/</pidfile> -->

    <!-- metadata used for stream listing -->
        <name>Asterisk Stream</name>
        <genre>Audio Streams</genre>

    <!--    Input module.

        This example uses the 'oss' module. It takes input from the
        OSS audio device (e.g. line-in), and processes it for live
        encoding.  -->
        <param name="rate">8000</param>
        <param name="channels">1</param>
        <!-- Read metadata (from stdin by default, or -->
        <!-- filename defined below (if the latter, only on SIGUSR1) -->
        <param name="metadata">1</param>
        <param name="metadatafilename">test</param>

    <!--    Stream instance.

        You may have one or more instances here.  This allows you to
        send the same input data to one or more servers (or to different
        mountpoints on the same server). Each of them can have different
        parameters. This is primarily useful for a) relaying to multiple
        independent servers, and b) encoding/reencoding to multiple

        If one instance fails (for example, the associated server goes
        down, etc), the others will continue to function correctly.
        This example defines a single instance doing live encoding at
        low bitrate.  -->

        <!--    Server details.

            You define hostname and port for the server here, along
            with the source password and mountpoint.  -->

        <yp>1</yp>   <!-- allow stream to be advertised on YP, default 0 -->

        <!--    Live encoding/reencoding:

            channels and samplerate currently MUST match the channels
            and samplerate given in the parameters to the oss input
            module above or the remsaple/downmix section below.  -->


        <!-- stereo->mono downmixing, enabled by setting this to 1 -->

        <!-- resampling.

            Set to the frequency (in Hz) you wish to resample to, -->

        <!-- <resample>
        </resample> -->



I just found it. Thanks!

I have run:
#/etc/init.d/asterisk start

When I dial in xxxx via phone, it’s not connected. If I run openam, I can record message. Is there any settings of asterisk need be done first to take incoming phone calls?

Sorry for my dummy question. I am new to the society.