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Hi, <br>
<br>
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?<br>
It should be like <a class="moz-txt-link-freetext" href="http://">http://</a><ip>:<port>/mcu <br>
<br>
Best regards<br>
Sergio<br>
<br>
Gonzalo Merayo escribió:
<blockquote
cite="mid:5e6cfcce0805021053u649ffe66mabe453620e365876@mail.gmail.com"
type="cite">
<pre wrap="">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 <a class="moz-txt-link-abbreviated" href="mailto:sip:8001@10.53.5.107">sip:8001@10.53.5.107</a> SIP/2.0
From: "Prueba"<a class="moz-txt-link-rfc2396E" href="mailto:sip:5555@10.53.5.16"><sip:5555@10.53.5.16></a>;tag=as51707987
User-Agent: Asterisk PBX
Date: Fri, 02 May 2008 14:43:50 GMT
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:8001@10.53.5.107"><sip:8001@10.53.5.107></a>
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: <a class="moz-txt-link-rfc2396E" href="mailto:sip:5555@10.53.5.16"><sip:5555@10.53.5.16></a>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Call-Id: <a class="moz-txt-link-abbreviated" href="mailto:63e315b74f0b4b3d18c3c8922dd7cac1@10.53.5.16">63e315b74f0b4b3d18c3c8922dd7cac1@10.53.5.16</a>
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
<a class="moz-txt-link-rfc2396E" href="mailto:sergio.garcia@fontventa.com"><sergio.garcia@fontventa.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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:
</pre>
<blockquote type="cite">
<pre wrap="">Hi,
</pre>
</blockquote>
<pre wrap=""> > 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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<pre wrap=""><!---->
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