[Asterisk-video] mcuWeb 404 not found

Sergio Garcia Murillo sergio.garcia at fontventa.com
Mon Oct 8 03:12:04 CDT 2012


Hi Gonzalo,

You are using and old version of the mcuWeb, which is not compatible 
with latest mediamixer.

Please, download also the latest binary from mcuWeb and try again.

Best regards
Sergio

El 07/10/2012 7:50, Gonzalo Gasca Meza escribió:
>
> Hi all,
> Not able to place new calls into McuWeb I get 404 Not found at SIP level
> Call flow:
>
> Jitsi <--> Asterisk 1.8 <--> McuWeb/Mcu
>
>
> Obtained logs and I get:
>
> org.apache.xmlrpc.XmlRpcException: Format string requests 3 items from 
> array, but array has only 1 items.
>
>
> Versions:
>
>  1. mcuWeb.sar Rev 474
>  2. xmlrpc-c-1.16.35
>  3. Latest mcu/mediamixer build from sourceforge
>
>
> Sailfin exception:
> http://pastebin.com/5vVVaCeg
>
>
> I get 404 Not Found INVITE from Asterisk PBX contains m line with Video.
>
> McuWeb
> Media Mixers
> Linkqbox http://110.10.0.210:8090/mcu 110.10.0.210 110.10.0.210 Up and 
> running...
>
> Profile List
> Linkqbox at 2048:30:6 Linkqbox HD720P-2048Kbs 30fps
>
> AdHoc Conference Templates
> 200 High Linkqbox Linkqbox
>
>
> INVITE sip:200 at 110.10.0.210:5070 SIP/2.0
>
> Via: SIP/2.0/UDP 110.10.0.200:5060;branch=z9hG4bK6f43166f;rport
>
> Max-Forwards: 70
>
> From: "Extension 111" sip:111 at 110.10.0.200 
> <mailto:sip:111 at 110.10.0.200>;tag=as32707173
>
> To: sip:200 at 110.10.0.210:5070 <mailto:sip:200 at 110.10.0.210:5070>
>
> Contact:sip:111 at 110.10.0.200:5060 <mailto:sip:111 at 110.10.0.200:5060>
>
> Call-ID: 3017f15f51f910ac6364a6493836f4c6 at 110.10.0.200:5060
>
> CSeq: 102 INVITE
>
> User-Agent: FPBX-2.10.1(1.8.11.0)
>
> Date: Sun, 07 Oct 2012 05:09:52 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH
>
> Supported: replaces, timer
>
> Content-Type: application/sdp
>
> Content-Length: 358
>
> v=0
> o=root 1791433792 1791433792 IN IP4 110.10.0.200
> s=Asterisk PBX 1.8.11.0
> c=IN IP4 110.10.0.200
> b=CT:4000
> t=0 0
> m=audio 14438 RTP/AVP 9 0 8 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 10988 RTP/AVP 99
> a=rtpmap:99 H264/90000
> a=sendrecv
>
>
> MCU log
>
> 0xb7f9d90001349586406.002-Handler on /status
> 0xb7f9d90001349586406.002-Handler on /mediagateway
> 0xb7f9d90001349586406.002-Handler on /mcu
> 0xb7f9cb9001349586406.0020xb7f9d90001349586406.002-Handler on /jsr309
> -RTMP Server Thread 29866
> 0xb7f9d90001349586406.002-Handler on /events/mcu
> 0xb7f9d90001349586406.0020xb7f9cb9001349586406.002>Run RTMP Server 
> 0xbfeb607c
> -Handler on /events/jsr309
> 0xb7f9d90001349586406.002-Handler on /broadcaster
> 0xb7f9cb9001349586406.002-Accepting connections 1
> 0xb759bb9001349586418.034-Dispatching /mcu/mcu
> 0xb759bb9001349586418.034>ProcessRequest uri:/mcu/mcu
> 0xb759bb9001349586418.035-ProcessRequest method:CreateConference
> 0xb759bb9001349586418.035Error processing xml cmd "Fault occurred"
> 0xb759bb9001349586418.036<ProcessRequest time:1
>
>
>
> --
> _____________________________________________________________________
> -- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-video/attachments/20121008/bdbadd18/attachment-0001.htm>


More information about the asterisk-video mailing list