[Asterisk-video] A question about video clip playback

[Ramtin Amin]

And whit the same eyebeam
can you put in [general]
videosuport=yes
disallow=all
allow=ulaw
allow=allow
allow=h263
 
 
and in extensions.conf in ur default,
 
exten => 9876,1,Answer()
exten => 9876,2,echo()
 
and try to see once you pressed on "start" button for the video if you see urself up there
 


Date: Fri, 30 Jun 2006 17:21:36 +0200From: af.devlist at gmail.comTo: asterisk-video at lists.digium.comSubject: Re: [Asterisk-video] A question about video clip playbackHello again,I worked a bit on my problem, trying to get the video to work.First, I tried recording a video from my eyebeam client on Asterisk using the Record() application. I use the following dialplan:[general]static=yeswriteprotect=no[video]exten => 1000,1,Answer()exten => 1000,n,Wait(1)exten => 1000,n,Record(testmessage:h263)exten => 1000,n,Hangup()my sip.conf is :[general]bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)bindaddr=0.0.0.0                ; IP address to bind to ( 0.0.0.0 binds to all)videosupport=yes[antoine]type=friendvideosupport=yessecret=antoinecallerid="antoine"host=dynamiccontext=videodisallow=allallow=gsmallow=h263 dtmfmode=rfc2833canreinvite=noI explicitly loaded the format_h263.so module in modules.conf I get eyebeam to send a correct SDP announcement, saying this (SDP contained in the 200 OK from Asterisk to eyebeam): v=0o=root 16577 16577 IN IP4 172.18.141.25s=sessionc=IN IP4 172.18.141.25b=CT:384t=0 0m=audio 10830 RTP/AVP 3a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - -a=sendrecvm=video 18812 RTP/AVP 34a=rtpmap:34 H263/90000a=sendrecvIn Asterisk I get the following error message :    -- Executing [1000 at video:1] Answer("SIP/antoine-4225", "") in new stack     -- Executing [1000 at video:2] Wait("SIP/antoine-4225", "1") in new stack    -- Executing [1000 at video:3] Record("SIP/antoine-4225", "testmessage:h263") in new stack    -- Playing 'beep' (language 'en') Jun 30 15:56:58 WARNING[4422]: translate.c:265 ast_translator_build_path: No translator path from g723 to unknownJun 30 15:56:58 WARNING[4422]: file.c:193 ast_writestream: Unable to translate to format h263, source format gsm Jun 30 15:56:58 WARNING[4422]: app_record.c:276 record_exec: Problem writing frame  == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-4225'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:v=0o=root 16577 16577 IN IP4 172.18.141.25s=sessionc=IN IP4 172.18.141.25b=CT:384t=0 0m=audio 13678 RTP/AVP 3a=rtpmap:3 GSM/8000a=silenceSupp:off - - - - a=sendrecvm=video 15878 RTP/AVP 34a=rtpmap:34 H263/90000a=sendrecvSniffing the connection with Ethereal, I receive no RTP packet for the video.When trying to play the video alone, I get the following error :     -- Executing [1000 at video:2] Playback("SIP/antoine-de5e", "/etc/asterisk/video/test") in new stackJun 30 16:05:39 WARNING[4460]: file.c:557 ast_openstream_full: File /etc/asterisk/video/test does not exist in any format 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 directoryJun 30 16:05:39 WARNING[4460]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/antoine-de5e for /etc/asterisk/video/test     -- Executing [1000 at video:3] Hangup("SIP/antoine-de5e", "") in new stack  == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-de5e'Do you have any explanation ?Thank you very much for your time and help,Antoine
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