You probably built the app_transcode application with an ffmpeg lib that you downloaded and when it runs, it loads uses another library that may come with your system.
Check how many ffmpeg libs you have and which version you are using. The symbol that you mention is most probably from the libswscale.so (used to resize the pictures). Emmanuel http://www.ives.fr/ matteo crivellaro a écrit : > Hi, > my webcam (Linksys WVC54GCA) supports the rtsp mobile streaming (I think it > is 3gp). > A part of my dialplan is: > > > [asgard] > exten=>goblin,1,h324m_gw_answer() > exten=>goblin,n,transcode(,s...@camera1,h...@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50) > > [camera1] > exten => s,1,Answer > exten => s,2,rtsp(rtsp://20.1.1.2:554/12345) > exten => s,3,HangUp > > > where 12345 is the code required by webcam and 554 is the rtsp port. > > When I start a video call appear the following errors: > > WARNING: app_rtsp.c:1037 rtsp_play: >rtsp play > WARNING: app_transcoder.c:963 > app_transcode:>Transcoding[,s...@camera1,h...@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50] > WARNING: channel.c:2781 set_format: Unable to find a codec translation path > form unknown to unknown > asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_transcoder.so: > undefined synbol: sws_getContext > > Can you help me? > > Regards, Matteo. > > > > > > > Matteo Crivellaro > Cassetta > e-mail: > [email protected] > 042653823 > C. Battisti 13/B > Cavarzere (Ve) Italy > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
