[Asterisk-video] dialing to mcuWeb

Fri May 2 12:53:24 CDT 2008

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
>  >
>  > _______________________________________________
>  > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>  >
>  > asterisk-video mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >    http://lists.digium.com/mailman/listinfo/asterisk-video
>  >
>
>
>
>  _______________________________________________
>  --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
>  asterisk-video mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video
>