[eliasj]

i configured the asterisk and the mcuweb, but when i try to make a videocall, it appears:

in the mcu when i make a call:

[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]>CreateParticipant
[29654]>CreateMixer video [1]
[29654]-Slots [0[29654]]
[29654]-Pos [1[29654]]
[29654]<CreateMixer video
[29654]>CreateMixer audio [1]
[29654]<CreateMixer audio
[29654]-SetVideoCodec [103,300,5,4,8]
[29654]-SetAudioCodec [3]
[29654]>Init video stream
[29654]>Init RTPSession
[29654]<Init RTPSession
[29654]<Init video stream
[29654]>Init audio stream
[29654]>Init RTPSession
[29654]<Init RTPSession
[29654]<Init audio stream
[29654]>Init mixer [1]
[29654]PipeVideoInput init
[29654]PipeVideoOutput init
[29654]<Init mixer [1]
[29654]>Init mixer [1]
[29654]PipeAudioOutput init
[29654]<Init mixer [1]
[29654]<CreateParticipant [1]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-SetAudioCodec [1]
[29654]-SetAudioCodec [8]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-SetVideoCodec[1]
[29654]-SetVideoCodec [103,10,128,4,12]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-StartSendingAudio [1]
[29654]>StartSending audio [201.48.87.17,15660]
[29654]-SetRemotePort [201.48.87.17,15660,8]
[29654]<StartSending audio [1]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]SendAudioThread [29654]
[29654]>SendAudio
[29654]-CreateAudioCodec [8]
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-StartSendingVideo [1]
[29654]>StartSending video [201.48.87.17,16448,103]
[29654]-SetRemotePort [201.48.87.17,16448,103]
[29654]<StartSending video [1]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]SendVideoThread [29654]
[29654]>SendVideo [176,144,128,10,4,12]
[29654]-CreateVideoEncoder [103,4,12]
[29654]-StartVideoCapture [176,144,10]
[29654]-SetSize [176,144]
[29654]-OpenCodec H263 [131072bps,10fps]
[29654]-Sending video
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-StartReceivingAudio [1]
[29654]<StartReceiving audio [36082]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]RecvAudioThread [29654]
[29654]>RecAudio
[29654]<ProccessRequest
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-StartReceivingVideo [1]
[29654]-StartReceiving Video [36472]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]RecVideoThread [29654]
[29654]>RecVideo
[29654]<ProccessRequest
[29654]-CreateAudioCodec [0]
[29654].[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0]
[29654]Error recv video [0] --------- this part is repeated until I close the call
[29654]-Dispatching [/mcu]
[29654]>ProcessRequest [/mcu]
[29654]>GetConferenceRef [100]
[29654]<GetConferenceRef
[29654]-DeleteParticipant [1]
[29654]-StopVideoCapture
[29654]<SendVideo [1]
[29654]>End
[29654]>StopSending [1]
[29654]Joining thread [145193872]
[29654]Joined [145193872]
[29654]<StopSending
[29654]>StopReceiving
[29654]Joining thread [-1219499120]
[29654]Error recv video [0]
[29654]<RecVideo
[29654]Joined [-1219499120]
[29654]<StopReceiving
[29654]<End
[29654]>StopSending Audio
[29654]-SendAudio cleanup[0]
[29654]-Deleting codec
[29654]<SendAudio
[29654]<StopSending Audio
[29654]>StopReceiving Audio
[29654]Silence
[29654]<RecAudio
[29654]<StopReceiving Audio
[29654]-DeleteMixer video [1]
[29654]>CalculatePositions
[29654]-Slots [0[29654]]
[29654]-Pos [0[29654]]
[29654]<CalculatePositions
[29654]-DeleteMixer audio [1]
[29654]>ReleaseConferenceRef [100]
[29654]<ReleaseConferenceRef
[29654]<ProccessRequest



and in the log of sailfin:


[#|2009-06-24T12:12:54.974-0400|WARNING|sun-comms-appserver1.0|global|_ThreadID=25;_ThreadName=SipContainer-serversWorkerThread-5060-1;_RequestID=5dafdbab-e149-4a7e-a5cf-ea94fd78730a;|sessionCreated!|#]

[#|2009-06-24T12:12:55.898-0400|WARNING|sun-comms-appserver1.0|global|_ThreadID=25;_ThreadName=SipContainer-serversWorkerThread-5060-1;_RequestID=5dafdbab-e149-4a7e-a5cf-ea94fd78730a;|SimpleProxyServlet: Got request:
INVITE sip:123@201.48.87.19 SIP/2.0
Content-Length: 366
To: <sip:123@201.48.87.19>
Contact: <sip:3499793418@201.48.87.17>
Supported: replaces
Cseq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Jun 2009 05:10:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Via: SIP/2.0/UDP 201.48.87.17:5060;rport=5060;branch=z9hG4bK57ff35db;received=201.48.87.17
Content-Type: application/sdp
From: "3499793418"<sip:3499793418@201.48.87.17>;tag=as015682aa
Call-Id: 4fa995f452649481116d4b7a6f0a43dd@201.48.87.17

v=0
o=root 10566 10566 IN IP4 201.48.87.17
s=session
c=IN IP4 201.48.87.17
b=CT:384
t=0 0
m=audio 15660 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16448 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
a=sendrecv
|#]


in the phone only appears "waiting for video".

the asterisk server is 201.48.87.17

the sailfin on 201.48.87.19 (i don't change the configuration)

the mcu is running on 201.48.87.19:8585


Anybody know how to resolve????

[erotavlas_turbo]