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<body>And whit the same eyebeam<BR>
can you put in [general]<BR>
videosuport=yes<BR>
disallow=all<BR>
allow=ulaw<BR>
allow=allow<BR>
allow=h263<BR>
<BR>
<BR>
and in <FONT class="">extensions.conf in ur default,</FONT><BR>
<BR>
exten => 9876,1,Answer()<BR>
exten => 9876,2,echo()<BR>
<BR>
and try to see once you pressed on "start" button for the video if you see urself up there<BR>
<BR ><BR ><BR ><BR ><BR > <BR>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #008080 2px solid; MARGIN-RIGHT: 0px">
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Date: Fri, 30 Jun 2006 17:21:36 +0200<BR>From: af.devlist@gmail.com<BR>To: asterisk-video@lists.digium.com<BR>Subject: Re: [Asterisk-video] A question about video clip playback<BR><BR>Hello again,<BR><BR>I worked a bit on my problem, trying to get the video to work.<BR><BR>First, I tried recording a video from my eyebeam client on Asterisk using the Record() application. I use the following dialplan:<BR><BR>[general]<BR>static=yes<BR>writeprotect=no<BR><BR>[video]<BR><BR>exten => 1000,1,Answer()<BR><BR>exten => 1000,n,Wait(1)<BR>exten => 1000,n,Record(testmessage:h263)<BR>exten => 1000,n,Hangup()<BR><BR>my sip.conf is :<BR><BR>[general]<BR>bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)<BR>bindaddr=<A href="http://0.0.0.0/" target=_blank>0.0.0.0</A> ; IP address to bind to (<A href="http://0.0.0.0/" target=_blank> 0.0.0.0</A> binds to all)<BR>videosupport=yes<BR><BR>[antoine]<BR>type=friend<BR>videosupport=yes<BR>secret=antoine<BR>callerid="antoine"<BR>host=dynamic<BR>context=video<BR>disallow=all<BR>allow=gsm<BR>allow=h263 <BR>dtmfmode=rfc2833<BR>canreinvite=no<BR><BR><BR>I explicitly loaded the format_h263.so module in modules.conf <BR><BR>I get eyebeam to send a correct SDP announcement, saying this (SDP contained in the 200 OK from Asterisk to eyebeam): <BR><BR>v=0<BR>o=root 16577 16577 IN IP4 <A href="http://172.18.141.25/" target=_blank>172.18.141.25</A><BR>s=session<BR>c=IN IP4 <A href="http://172.18.141.25/" target=_blank>172.18.141.25</A><BR>b=CT:384<BR>t=0 0<BR>m=audio 10830 RTP/AVP 3<BR>a=rtpmap:3 GSM/8000 <BR>a=silenceSupp:off - - - -<BR>a=sendrecv<BR>m=video 18812 RTP/AVP 34<BR>a=rtpmap:34 H263/90000<BR>a=sendrecv<BR><BR>In Asterisk I get the following error message :<BR><BR> -- Executing [1000@video:1] Answer("SIP/antoine-4225", "") in new stack <BR> -- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new stack<BR> -- Executing [1000@video:3] Record("SIP/antoine-4225", "testmessage:h263") in new stack<BR> -- Playing 'beep' (language 'en') <BR>Jun 30 15:56:58 WARNING[4422]: translate.c:265 ast_translator_build_path: No translator path from g723 to unknown<BR>Jun 30 15:56:58 WARNING[4422]: file.c:193 ast_writestream: Unable to translate to format h263, source format gsm <BR>Jun 30 15:56:58 WARNING[4422]: app_record.c:276 record_exec: Problem writing frame<BR> == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-4225'<BR><BR>I performed other tests trying to play test.h263 encoded with ffmpeg and test.gsm containing the subsequent audio track. They were built from the same video file, and are the same length. I just have the sound track, while the SDP message stipulates:<BR><BR>v=0<BR>o=root 16577 16577 IN IP4 <A href="http://172.18.141.25/" target=_blank>172.18.141.25</A><BR>s=session<BR>c=IN IP4 <A href="http://172.18.141.25/" target=_blank>172.18.141.25</A><BR>b=CT:384<BR>t=0 0<BR>m=audio 13678 RTP/AVP 3<BR>a=rtpmap:3 GSM/8000<BR>a=silenceSupp:off - - - - <BR>a=sendrecv<BR>m=video 15878 RTP/AVP 34<BR>a=rtpmap:34 H263/90000<BR>a=sendrecv<BR><BR>Sniffing the connection with Ethereal, I receive no RTP packet for the video.<BR><BR>When trying to play the video alone, I get the following error : <BR><BR> -- Executing [1000@video:2] Playback("SIP/antoine-de5e", "/etc/asterisk/video/test") in new stack<BR>Jun 30 16:05:39 WARNING[4460]: file.c:557 ast_openstream_full: File /etc/asterisk/video/test does not exist in any format <BR>Jun 30 16:05:39 WARNING[4460]: file.c:810 ast_streamfile: Unable to open /etc/asterisk/video/test (format 0x80002 (gsm|h263)): No such file or directory<BR>Jun 30 16:05:39 WARNING[4460]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/antoine-de5e for /etc/asterisk/video/test <BR> -- Executing [1000@video:3] Hangup("SIP/antoine-de5e", "") in new stack<BR> == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-de5e'<BR><BR><BR>Do you have any explanation ?<BR><BR>Thank you very much for your time and help,<BR><BR>Antoine<BR><BR><BR></BLOCKQUOTE><br /><hr />Le futur Hotmail : <a href='http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d' target='_new'>Essayez Windows Live Mail Beta</a></body>
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