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[Asterisk-video] app_rtsp

 
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geisj at pagestation.com
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PostPosted: Thu Apr 30, 2009 2:47 pm    Post subject: [Asterisk-video] app_rtsp

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@smvoice-sip:1] Answer("SIP/440-b4000900", "") in
new stack
-- Executing [50@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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geisj at pagestation.com
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PostPosted: Fri May 01, 2009 5:35 pm    Post subject: [Asterisk-video] app_rtsp

Seems like app_rtsp doesnt handle user name and password in the normal
manner.

I specified rtsp(rtsp://user:passwd@192.168.1.8/img/video.sav)
and it gives me an error.

I can do mplayer rtsp://user:passwd@192.168.1.8/img/video.sav and all is
correct.

Is there a different way to specify it?

Jerry

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