Docker Launcher cannot start when enabling API Manager in Server.xml

Describe the bug When using the docker launcher with the 0.15.13 build, the launcher cannot start a docker container when Server.xml enables API manager.
To Reproduce
Server.xml:
<?xml version="1.0" encoding="UTF-8"?>
<Server version="8">
<Name>OvenMediaEngine</Name>
<!-- Host type (origin/edge) -->
<Type>origin</Type>
<!-- Specify IP address to bind ("*" means all IPv4 IPs, "::" means all IPv6 IPs) -->
<!-- Uncomment the line below to enable IPv6 -->
<!-- <IP>::</IP> -->
<IP>*</IP>
<PrivacyProtection>false</PrivacyProtection>

<!-- 
To get the public IP address(mapped address of stun) of the local server. 
This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment. 
If this is successful, you can use ${PublicIP} in your settings.
-->
<StunServer>stun.ovenmediaengine.com:13478</StunServer>

<Modules>
    <!-- 
    Currently OME only supports h2 like all browsers do. Therefore, HTTP/2 only works on TLS ports.         
    -->
    <HTTP2>
        <Enable>true</Enable>
    </HTTP2>

    <LLHLS>
        <Enable>true</Enable>
    </LLHLS>

    <!-- P2P works only in WebRTC and is experiment feature -->
    <P2P>
        <!-- disabled by default -->
        <Enable>false</Enable>
        <MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
    </P2P>
</Modules>

<!-- Settings for the ports to bind -->
<Bind>
    <Managers>
        <API>
            <Port>8081</Port>
            <TLSPort>8082</TLSPort>
            <WorkerCount>1</WorkerCount>
        </API>
    </Managers>

    <Providers>
        <!-- Pull providers -->
        <RTSPC>
            <WorkerCount>1</WorkerCount>
        </RTSPC>
        <OVT>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <!-- Push providers -->
        <RTMP>
            <Port>1935</Port>
            <WorkerCount>1</WorkerCount>
        </RTMP>
        <SRT>
            <Port>9999</Port>
            <WorkerCount>1</WorkerCount>
        </SRT>
        <MPEGTS>
            <!--
                Below is an example of listening on port 4000-4005 (<Port>4000-4004,4005/udp</Port>)
                This is just a demonstration to show that you can configure the port in several ways

                <Port>4000-4004,4005/udp</Port>
            -->
            <Port>4000/udp</Port>
        </MPEGTS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>

            <IceCandidates>
                <!-- Uncomment the line below to use IPv6 ICE Candidate -->
                <!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
                <IceCandidate>*:10000-10004/udp</IceCandidate>

                <!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
                <!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->

                <!-- 
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Providers>

    <Publishers>
        <OVT>
            <Port>9000</Port>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <LLHLS>
            <!-- 
            OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports. 
            LLHLS works with higher performance over HTTP/2, 
            so it is recommended to use a TLS port.
            -->
            <Port>3333</Port>
            <!-- If you want to use TLS, specify the TLS port -->
            <TLSPort>3334</TLSPort>
            <WorkerCount>1</WorkerCount>
        </LLHLS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>
            <IceCandidates>
                <!-- Uncomment the line below to use IPv6 ICE Candidate -->
                <!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
                <IceCandidate>*:10000-10004/udp</IceCandidate>

                <!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
                <!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->

                <!-- 
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Publishers>
</Bind>
            <Managers>
            <Host>
                    <Names>
                            <Name>*</Name>
                    </Names>
            </Host>
            <API>
                    <AccessToken>your_access_token</AccessToken>
            </API>
    </Managers>

<!--
    Refer https://airensoft.gitbook.io/ovenmediaengine/alert
<Alert>
    <Url></Url>
    <SecretKey></SecretKey>
    <Timeout>3000</Timeout>
    <Rules>
        <Ingress>
            <MinBitrate></MinBitrate>
            <MaxBitrate></MaxBitrate>
            <MinFramerate></MinFramerate>
            <MaxFramerate></MaxFramerate>
            <MinWidth></MinWidth>
            <MinHeight></MinHeight>
            <MaxWidth></MaxWidth>
            <MaxHeight></MaxHeight>
            <MinSamplerate></MinSamplerate>
            <MaxSamplerate></MaxSamplerate>
            <LongKeyFrameInterval />
            <HasBFrames />
        </Ingress>
    </Rules>
</Alert>
-->

<VirtualHosts>
    <!-- You can use wildcard like this to include multiple XMLs -->
    <VirtualHost include="VHost*.xml" />
    <VirtualHost>
        <Name>default</Name>
        <!--Distribution is a value that can be used when grouping the same vhost distributed across multiple servers. This value is output to the events log, so you can use it to aggregate statistics. -->
        <Distribution>ovenmediaengine.com</Distribution>

        <!-- Settings for multi ip/domain and TLS -->
        <Host>
            <Names>
                <!-- Host names
                    <Name>stream1.airensoft.com</Name>
                    <Name>stream2.airensoft.com</Name>
                    <Name>*.sub.airensoft.com</Name>
                    <Name>192.168.0.1</Name>
                -->
                <Name>*</Name>
            </Names>
            <!--
            <TLS>
                <CertPath>path/to/file.crt</CertPath>
                <KeyPath>path/to/file.key</KeyPath>
                <ChainCertPath>path/to/file.crt</ChainCertPath>
            </TLS>
            -->
        </Host>

        <!--    
        Refer https://airensoft.gitbook.io/ovenmediaengine/signedpolicy
        <SignedPolicy>
            <PolicyQueryKeyName>policy</PolicyQueryKeyName>
            <SignatureQueryKeyName>signature</SignatureQueryKeyName>
            <SecretKey>aKq#1kj</SecretKey>

            <Enables>
                <Providers>rtmp,webrtc,srt</Providers>
                <Publishers>webrtc,hls,llhls,dash,lldash</Publishers>
            </Enables>
        </SignedPolicy>
        -->

        <!--
        <AdmissionWebhooks>
            <ControlServerUrl></ControlServerUrl>
            <SecretKey></SecretKey>
            <Timeout>3000</Timeout>
            <Enables>
                <Providers>rtmp,webrtc,srt</Providers>
                <Publishers>webrtc,hls,llhls,dash,lldash</Publishers>
            </Enables>
        </AdmissionWebhooks>
        -->

        <!-- <Origins>
            <Properties>
                <NoInputFailoverTimeout>3000</NoInputFailoverTimeout>
                <UnusedStreamDeletionTimeout>60000</UnusedStreamDeletionTimeout>
            </Properties>
            <Origin>
                <Location>/app/stream</Location>
                <Pass>
                    <Scheme>ovt</Scheme>
                    <Urls><Url>origin.com:9000/app/stream_720p</Url></Urls>
                </Pass>
                <ForwardQueryParams>false</ForwardQueryParams>
            </Origin>
            <Origin>
                <Location>/app/</Location>
                <Pass>
                    <Scheme>ovt</Scheme>
                    <Urls><Url>origin.com:9000/app/</Url></Urls>
                </Pass>
            </Origin>
            <Origin>
                <Location>/edge/</Location>
                <Pass>
                    <Scheme>ovt</Scheme>
                    <Urls><Url>origin.com:9000/app/</Url></Urls>
                </Pass>
            </Origin>
        </Origins> -->

        <!--
        <OriginMapStore>
            In order to use OriginMap, you must enable OVT Publisher in Origin and OVT Provider in Edge.
            <RedisServer>
                <Host>192.168.0.160:6379</Host>
                <Auth>!@#ovenmediaengine</Auth>
            </RedisServer>

            This is only needed for the origin server and used to register the ovt address of the stream. 
            <OriginHostName>ome-dev.airensoft.com</OriginHostName>
        </OriginMapStore>
        -->

        <!-- Default CORS Settings -->
        <CrossDomains>
            <Url>*</Url>
        </CrossDomains>

        <!-- Settings for applications -->
        <Applications>
            <Application>
                <Name>app</Name>
                <!-- Application type (live/vod) -->
                <Type>live</Type>
                <OutputProfiles>
                    <!-- Enable this configuration if you want to hardware acceleration using GPU -->
                    <HardwareAcceleration>false</HardwareAcceleration>
                    <OutputProfile>
                        <Name>bypass_stream</Name>
                        <OutputStreamName>${OriginStreamName}</OutputStreamName>

                        <!-- 
                        You can provide ABR with Playlist. Currently, ABR is supported in LLHLS and WebRTC.

                        <Playlist>
                            <Name>Paid</Name>
                            You can play this playlist with 
                            LLHLS : http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8 
                            WebRTC : ws[s]://<domain>[:port]/<app>/<stream>/<FileName>
                            Note that the keywords "playlist" and "chunklist" MUST NOT be included in FileName.
                            <FileName>premium</FileName>

                            Options is optional. 
                            <Options>
                                WebRtcAutoAbr : Default value is true
                                <WebRtcAutoAbr>true</WebRtcAutoAbr> 
                            </Options>

                            <Rendition>
                                <Name>bypass</Name>
                                <Video>bypass_video</Video>
                                <Audio>bypass_audio</Audio>
                            </Rendition>
                            <Rendition>
                                <Name>480p</Name>
                                <Video>480p</Video>
                                <Audio>bypass_audio</Audio>
                            </Rendition>
                            <Rendition>
                                <Name>720p</Name>
                                <Video>720p</Video>
                                <Audio>bypass_audio</Audio>
                            </Rendition>
                        </Playlist>
                        <Playlist>
                            <Name>free</Name>
                            <FileName>free</FileName>
                            <Rendition>
                                <Name>720p</Name>
                                <Video>720p</Video>
                                <Audio>bypass_audio</Audio>
                            </Rendition>
                        </Playlist>
                        -->

                        <Encodes>
                            <Video>
                                <Name>bypass_video</Name>
                                <Bypass>true</Bypass>
                            </Video>
                            <Audio>
                                <Name>aac_audio</Name>
                                <Codec>aac</Codec>
                                <Bitrate>128000</Bitrate>
                                <Samplerate>48000</Samplerate>
                                <Channel>2</Channel>
                                <BypassIfMatch>
                                    <Codec>eq</Codec>
                                </BypassIfMatch>
                            </Audio>
                            <Audio>
                                <Name>opus_audio</Name>
                                <Codec>opus</Codec>
                                <Bitrate>128000</Bitrate>
                                <Samplerate>48000</Samplerate>
                                <Channel>2</Channel>
                                <BypassIfMatch>
                                    <Codec>eq</Codec>
                                </BypassIfMatch>
                            </Audio>
                            <!--
                            <Video>
                                <Name>video_1280</Name>
                                <Codec>h264</Codec>
                                <Bitrate>5024000</Bitrate>
                                <Width>1920</Width>
                                <Height>1280</Height>
                                <Framerate>30</Framerate>
                                <KeyFrameInterval>30</KeyFrameInterval>
                                <BFrames>0</BFrames>
                                <Preset>faster</Preset>
                            </Video>
                            <Video>
                                <Name>video_720</Name>
                                <Codec>h264</Codec>
                                <Bitrate>2024000</Bitrate>
                                <Width>1280</Width>
                                <Height>720</Height>
                                <Framerate>30</Framerate>
                                <KeyFrameInterval>30</KeyFrameInterval>
                                <BFrames>0</BFrames>                                    
                                <Preset>faster</Preset>
                            </Video>    
                            <Audio>
                                <Name>bypass_audio</Name>
                                <Bypass>true</Bypass>
                            </Audio>                                                                
                            -->
                            <!-- 
                                 If all conditions of BypassIfMatch are matched, bypass is used without encoding.
                                 [Options] 
                                   eq: euqal to
                                   lte: less than or equal to
                                   gte: greater than or equal to

                            <Video>
                                <Name>conditional_video_encoding</Name>
                                <Codec>h264</Codec>
                                <Bitrate>2024000</Bitrate>
                                <Width>1280</Width>
                                <Height>720</Height>
                                <Framerate>30</Framerate>
                                <BypassIfMatch>
                                    <Codec>eq</Codec>
                                    <Framerate>lte</Framerate>
                                    <Width>lte</Width>
                                    <Height>lte</Height>
                                    <SAR>eq</SAR>
                                </BypassIfMatch>                                    
                            </Video>
                            <Audio>
                                <Name>conditional_audio_encoding</Name>
                                <Codec>aac</Codec>
                                <Bitrate>128000</Bitrate>
                                <Samplerate>48000</Samplerate>
                                <Channel>2</Channel>
                                <BypassIfMatch>
                                    <Codec>eq</Codec>
                                    <Samplerate>lte</Samplerate>
                                    <Channel>eq</Channel>
                                </BypassIfMatch>
                            </Audio>                                                                            
                            -->
                        </Encodes>
                    </OutputProfile>
                </OutputProfiles>
                <Providers>
                    <OVT />
                    <WebRTC />
                    <RTMP />
                    <SRT />
                    <MPEGTS>
                        <StreamMap>
                            <!--
                                Set the stream name of the client connected to the port to "stream_${Port}"
                                For example, if a client connects to port 4000, OME creates a "stream_4000" stream
                                <Stream>
                                    <Name>stream_${Port}</Name>
                                    <Port>4000,4001-4004</Port>
                                </Stream>
                                <Stream>
                                    <Name>stream_4005</Name>
                                    <Port>4005</Port>
                                </Stream>
                            -->
                            <Stream>
                                <Name>stream_${Port}</Name>
                                <Port>4000</Port>
                            </Stream>
                        </StreamMap>
                    </MPEGTS>
                    <RTSPPull />
                    <WebRTC>
                        <Timeout>30000</Timeout>
                        <CrossDomains>
                            <Url>*</Url>
                        </CrossDomains>
                    </WebRTC>
                </Providers>
                <Publishers>
                    <AppWorkerCount>1</AppWorkerCount>
                    <StreamWorkerCount>8</StreamWorkerCount>
                    <OVT />
                    <WebRTC>
                        <Timeout>30000</Timeout>
                        <Rtx>false</Rtx>
                        <Ulpfec>false</Ulpfec>
                        <JitterBuffer>false</JitterBuffer>
                    </WebRTC>
                    <LLHLS>
                        <ChunkDuration>0.5</ChunkDuration>
                        <!-- <PartHoldBack> SHOULD be at least three times the <ChunkDuration> -->
                        <PartHoldBack>1.5</PartHoldBack>
                        <SegmentDuration>6</SegmentDuration>
                        <SegmentCount>10</SegmentCount>
                        <DVR>
                            <Enable>false</Enable>
                            <TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
                            <MaxDuration>3600</MaxDuration>
                        </DVR>
                                                    <RTMPPush>
                                                    </RTMPPush>
                        <CrossDomains>
                            <Url>*</Url>
                        </CrossDomains>
                    </LLHLS>
                </Publishers>
            </Application>
        </Applications>
    </VirtualHost>
</VirtualHosts>
</Server>
Use ./ome_docker_launcher.sh start.
The Launcher throws an error.
Expected behavior A clear and concise description of what you expected to happen.
Logs • Starting OvenMediaEngine... • Obtaining the port list from /usr/share/ovenmediaengine/conf/Server.xml • ERROR: Not supported type: • This may be a bug of this launcher, and should not be occurred. • Please report this issue to https://github.com/AirenSoft/OvenMediaEngine/issues
Server (please complete the following information):
OS: Ubuntu 22.10
OvenMediaEngine Version: 0.15.13
Branch: main
Player (please complete the following information):
Device: N/A
OS: N/A
Browser N/A
Version N/A
Additional context It was frustrating as a newcomer to OverMediaEngine to battle this problem when it was much easier to just default back to docker in ubuntu.
AirenSoft / OvenMediaEngine

Docker Launcher cannot start when enabling API Manager in Server.xml #1306
