[Asterisk-video] app_rtsp

Jerry Geis geisj at pagestation.com
Thu Apr 30 10:39:13 CDT 2009


I have a GXP3000 and a linksys WVC210 camera (rtsp capable) mplayer 
plays rtsp from the camera.

asterisk is 1.4.23.2

I changed rtp.c [99] line copied it and changed to [126]

Now when my GXP calls the camera I get the following:

Any ideas?

Jerry
----------------------------------
    -- Executing [50 at smvoice-sip:1] Answer("SIP/440-b4000900", "") in 
new stack
    -- Executing [50 at smvoice-sip:2] rtsp("SIP/440-b4000900", 
"rtsp://192.168.1.175/img/video.sav") in new stack
[Apr 30 11:33:01] WARNING[8827]: app_rtsp.c:1037 rtsp_play: >rtsp play
[Apr 30 11:33:02] WARNING[8827]: channel.c:2930 set_format: Unable to 
find a codec translation path from 0x200004 (ulaw|h264) to 0x0 (nothing)
[Apr 30 11:33:02] ERROR[8827]: app_rtsp.c:1273 rtsp_play: No media found
[Apr 30 11:33:02] WARNING[8827]: app_rtsp.c:1535 rtsp_play: <rtsp_play  
== Auto fallthrough, channel 'SIP/440-b4000900' status is 'UNKNOWN'
    -- Saved useragent "Asterisk PBX" for peer devcentos5x64_to_nvidiawallrd
    -- Saved useragent "Asterisk PBX" for peer devcentos5x64_to_ebox4300cf
    -- Saved useragent "Asterisk PBX" for peer devcentos5x64_to_ebox4300
    -- Saved useragent "Asterisk PBX" for peer devcentos5x64_to_am2mm
    -- Saved useragent "Asterisk PBX SVN--r" for peer 
devcentos5x64_to_ebox3850
    -- Saved useragent "Asterisk PBX" for peer devcentos5x64_to_mmfirepanel





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