[Asterisk-video] dialing to mcuWeb

Fri May 2 14:14:54 CDT 2008

The mixer is running and url is: http://127.0.0.1:9999/mcu
When I create the conference it says:
[6184]-Dispatching [/mcu]
[6184]>ProcessRequest [/mcu]
[6184]>CreateConference
[6184]<CreateConferencei [100]
[6184]>GetConferenceRef [100]
[6184]<GetConferenceRef
[6184]-Init multiconf
[6184]-SetCompositionType [1,0]
[6184]>SetCompositionType [1,0]
[6184]>CalculatePositions
[6184]-Slots [0[6184],0[6184],0[6184],0[6184]]
[6184]-Pos   [0[6184],0[6184],0[6184],0[6184]]
[6184]<CalculatePositions
[6184]<SetCompositionType
[image2 @ 0xb795ec34]Could not find codec parameters (Video: png, yuv420p)
[6184]Couldn't find stream information for the logo image file...
[6184]>ReleaseConferenceRef [100]
[6184]<ReleaseConferenceRef
[6184]<ProccessRequest
[6184]-MixAudioThread [6184]
[6184]-MixVideoThread [6184]
[6184]>MixVideo

When I call it says:
[6184]-Dispatching [/mcu]
[6184]>ProcessRequest [/mcu]
[6184]>GetConferenceRef [100]
[6184]<GetConferenceRef
[6184]>CreateParticipant
[6184]>CreateMixer video [1]
[6184]-Slots [0[6184],0[6184],0[6184],0[6184]]
[6184]-Pos   [1[6184],0[6184],0[6184],0[6184]]
[6184]<CreateMixer video
[6184]>CreateMixer audio [1]
[6184]<CreateMixer audio
[6184]-SetVideoCodec [103,300,5,4,8]
[6184]-SetAudioCodec [3]
[6184]>Init video stream
[6184]<Init video stream
[6184]>Init audio stream
[6184]<Init audio stream
[6184]>Init mixer [1]
[6184]PipeVideoInput init
[6184]PipeVideoOutput init
[6184]<Init mixer [1]
[6184]>Init mixer [1]
[6184]PipeAudioOutput init
[6184]<Init mixer [1]
[6184]<CreateParticipant [1]
[6184]>ReleaseConferenceRef [100]
[6184]<ReleaseConferenceRef
[6184]<ProccessRequest
[6184]-Dispatching [/mcu]
[6184]>ProcessRequest [/mcu]
[6184]>GetConferenceRef [100]
[6184]<GetConferenceRef
[6184]-SetAudioCodec [1]
[6184]-SetAudioCodec [0]
[6184]>ReleaseConferenceRef [100]
[6184]<ReleaseConferenceRef
[6184]<ProccessRequest

On Fri, May 2, 2008 at 3:47 PM, Sergio Garcia Murillo
<sergio.garcia at fontventa.com> wrote:
>
>  Hi,
>
>  Is the media mixer running? Which url have you configured in the mcuWeb for
> the mixer? What is the log of the mixer when you create the conference?
>  It should be like http://<ip>:<port>/mcu
>
>
>  Best regards
>  Sergio
>
>  Gonzalo Merayo escribió:
>  Hi Sergio,
>  Thanks, I got it dialing with Dial(SIP/AS/<did>).
>  Now it dials but doesn't pick up. The log says:
>
> [#|2008-05-02T10:41:43.553-0400|WARNING|sun-comms-appserver1.0|global|_ThreadID=49;_ThreadName=SipContainer-serversWorkerThread-5060-1;_RequestID=d065ae08-ab16-4292-a7f0-7a1d17683fb6;|sessionCreated!|#]
>
> [#|2008-05-02T10:41:43.600-0400|SEVERE|sun-comms-appserver1.0|global|_ThreadID=49;_ThreadName=SipContainer-serversWorkerThread-5060-1;_RequestID=d065ae08-ab16-4292-a7f0-7a1d17683fb6;|The
> log message is null.
> org.apache.xmlrpc.common.XmlRpcExtensionException: Null values aren't
> supported, if isEnabledForExtensions() == false
>  at
> org.apache.xmlrpc.common.TypeFactoryImpl.getSerializer(TypeFactoryImpl.java:115)
>  at
> org.apache.xmlrpc.serializer.XmlRpcWriter.writeValue(XmlRpcWriter.java:168)
>  at org.apache.xmlrpc.serializer.XmlRpcWriter.write(XmlRpcWriter.java:77)
>
> [#|2008-05-02T10:41:43.601-0400|WARNING|sun-comms-appserver1.0|global|_ThreadID=49;_ThreadName=SipContainer-serversWorkerThread-5060-1;_RequestID=d065ae08-ab16-4292-a7f0-7a1d17683fb6;|SimpleProxyServlet:
> Got request:
> INVITE sip:8001 at 10.53.5.107 SIP/2.0
> From: "Prueba"<sip:5555 at 10.53.5.16>;tag=as51707987
> User-Agent: Asterisk PBX
> Date: Fri, 02 May 2008 14:43:50 GMT
> To: <sip:8001 at 10.53.5.107>
> Content-Type: application/sdp
> Via: SIP/2.0/UDP
> 10.53.5.16:5060;received=10.53.5.16;branch=z9hG4bK28769221;rport=5060
> Max-Forwards: 70
> Content-Length: 258
> Cseq: 102 INVITE
> Contact: <sip:5555 at 10.53.5.16>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Call-Id: 63e315b74f0b4b3d18c3c8922dd7cac1 at 10.53.5.16
>
> v=0
> o=root 7091 7091 IN IP4 10.53.5.16
> s=session
> c=IN IP4 10.53.5.16
> t=0 0
> m=audio 16974 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> |#]
>
> And a little later:
>
> [#|2008-05-02T10:42:24.158-0400|WARNING|sun-comms-appserver1.0|javax.enterprise.system.container.sip|_ThreadID=45;_ThreadName=Thread-29;_RequestID=f33053f2-d508-4d23-84fe-1184e72362b2;|"Failed
> get read mutex"|#]
>
> My guessing is that it was unable to connect to mcu mixer server and
> thats why it didn't pick up.
> Do you recomend I try it with the next binary distro?
> I've realized the wiki at mcu has not been used. do you mind if I copy
> these findings on it? is there another wiki or doc source up to date?
>
> Thanks
>
> On Fri, May 2, 2008 at 12:46 PM, Sergio Garcia Murillo
> <sergio.garcia at fontventa.com> wrote:
>
>
>  Hi Gonzalo
>
>  First you have to create a conference in the mcuWeb interface and
>  assigng a did number, then just set up your dialplan to match the did
>  like this:
>
>  exten => 8000,1,Dial(SIP/<did>@AS)
>
>  I've made some improvements both the mcuWeb and in the media mixer, I'll
>  commit them during the weekend and upload a new compiled sar to the
>  web.
>
>  Best regards
>  Sergio
>
>
>  Gonzalo Merayo escribió:
>
>
>
>
>  Hi,
>
>  > Thanks Sergio, I got the correct version of xmlrpc and got
>  > everithing compiled. Now I'm trying to connect to a conference.
>  > When I route the call to mcuWeb the log says:
>  >
>  >
> [#|2008-05-02T07:18:40.059-0400|SEVERE|sun-comms-appserver1.0|javax.enterprise.system.container.sip|_ThreadID=47;_ThreadName=SipContainer-serversWorkerThread-5060-6;_RequestID=f5f56ae7-0fbd-4ddc-aec8-cad8090c3ba9;|Exception
>  > allocating servlet
>  > java.lang.NullPointerException
>  > at
> org.murillo.mcuWeb.ConferenceMngr.getMappedConference(ConferenceMngr.java:174)
>  > at org.murillo.mcuWeb.MCUSipServlet.doInvite(MCUSipServlet.java:77)
>  > at javax.servlet.sip.SipServlet.doRequest(SipServlet.java:54)
>  > at javax.servlet.sip.SipServlet.service(SipServlet.java:43)
>  >
>  > That line is:
>  > if(uri.equals(conf.getDID()))
>  > most likely the null object is uri since conf is used just above.
>  >
>  > I'm guessing Asterisk is not sending some parameter when connecting to
>  > mcu, but I was unable to find which one.
>  > My dial rule is:
>  > exten => 8000,1,Dial(SIP/AS)
>  >
>  > When I try something like
>  > exten => 8000,1,Dial(SIP/AS/555)
>  > mcu fails with incorrect paramete
>  >
>  > Does anyone know what I'm doing wrong?
>  >
>  > Thanks
>  > Regards
>  >
>  > Gonzalo
>  >
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