Hi this is due to codec amr is not properly installed
On Fri, Feb 25, 2011 at 6:06 PM, Mário Dias <ma...@hardserver.com> wrote: > Hello agian! > > I forgot another error in asterisk logs: > > [Feb 25 18:03:30] WARNING[18705] app_transcoder.c: >Transcoding > [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,180004] > [Feb 25 18:03:30] WARNING[18707] app_rtsp.c: >rtsp play > [Feb 25 18:03:31] WARNING[18707] channel.c: Unable to find a codec > translation path from 0x780004 (ulaw|h263|h263p|h264) to 0x2000 > (nothing) > [Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received > from '192.168.0.89' > [Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received > from '192.168.0.89' > [Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received > from '192.168.0.89' > > What is the problem???? > > 2011/2/25 Mário Dias <ma...@hardserver.com>: > > Hello! I just try reinstall ffmpeg in other version of linux (ubuntu) > > and the before error not appear now. > > > > But, When I call 5001, the video call answer but not appear the video > > (waitting remote video) in X-lite4. > > > > In asterisk logs there are: > > > > [Feb 25 17:46:54] WARNING[18490] app_transcoder.c: >Transcoding > > [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,180004] > > [Feb 25 17:46:54] WARNING[18492] app_rtsp.c: >rtsp play > > [Feb 25 17:46:54] WARNING[18492] channel.c: Unable to find a codec > > translation path from 0x780004 (ulaw|h263|h263p|h264) to 0x2000 > > (nothing) > > > > why ??? > > > > I remember that I allowed in sip.conf : video support, h263, h263p, h264 > > > > I want transcode the codec of received video RTSP streaming (codec > > mp4v) to H263 of my softphone..... > > > > > > > > > > 2011/2/25 Mário Dias <ma...@hardserver.com>: > >> Sergio, > >> > >> The results of command ffmpeg -formats | grep h263 > >> > >> > >> asterisk2:/# ffmpeg -formats | grep h263 > >> FFmpeg version r11872+debian_0.svn20080206-18+lenny3, Copyright (c) > >> 2000-2008 Fabrice Bellard, et al. > >> configuration: --enable-gpl --enable-libfaad --enable-pp > >> --enable-swscaler --enable-x11grab --prefix=/usr --enable-libgsm > >> --enable-libtheora --enable-libvorbis --enable-pthreads > >> --disable-strip --enable-libdc1394 --disable-armv5te --disable-armv6 > >> --disable-altivec --disable-vis --enable-shared --disable-static > >> libavutil version: 49.6.0 > >> libavcodec version: 51.50.0 > >> libavformat version: 52.7.0 > >> libavdevice version: 52.0.0 > >> built on Feb 13 2011 03:56:05, gcc: 4.3.2 > >> DE h263 raw h263 > >> D VSDT h263 > >> D VSD h263i > >> even though both encoding and decoding are supported. For example, the > h263 > >> decoder corresponds to the h263 and h263p encoders, for file formats it > is even > >> > >> > >> and now?? What I have to do to solve my issue?? > >> > >> Best regards, > >> > >> Mário Dias > >> > >> > >>> 2011/2/24 Sergio Garcia Murillo <sergio.gar...@fontventa.com>: > >>>> The app_transcoder is loaded correctly: > >>>> > >>>> [Feb 23 18:15:22] ERROR[4142] app_transcoder.c: Error opening encoder > >>>> > >>>> Could you check if your libavcodec.so library supports h263 encoding? > >>>> > >>>>>ffmpeg -formats | grep h263 > >>>> DE h263 raw H.263 > >>>> > >>>> BR > >>>> Sergio > >>>> > >>>> El 24/02/2011 21:51, Mitul Limbani escribió: > >>>>> > >>>>> Hi Mario, > >>>>> > >>>>> Can you check if the app_transcoder.so got loaded without any problem > >>>>> within Asterisk Startup ? > >>>>> > >>>>> you can try this: > >>>>> > >>>>> core set verbose 5 > >>>>> module unload app_transcode.so > >>>>> module load app_transcode.so > >>>>> > >>>>> and paste the output. > >>>>> > >>>>> Regards, > >>>>> Mitul Limbani > >>>>> Enterux Solutions, > >>>>> www.enterux.com > >>>>> > >>>>> Quoting Mário Dias <ma...@hardserver.com>: > >>>>> > >>>>>> Hello! I just installed the app_transcoder with success and this > runs > >>>>>> well with asterisk boot... > >>>>>> > >>>>>> Now the problem is: > >>>>>> > >>>>>> My extensions.conf: > >>>>>> > >>>>>> [default] > >>>>>> > >>>>>> exten=5001,1,Answer() > >>>>>> > >>>>>> exten=5001,n,Transcode(,s@camera,h263@qcif > /fps=10/kb=52/qmin=4/qmax=12/gs=50) > >>>>>> exten=5001,n,Hangup() > >>>>>> > >>>>>> [camera] > >>>>>> > >>>>>> exten=s,1,Answer() > >>>>>> exten=s,n,Rtsp(rtsp://192.168.10.14:8554/CH001.sdp) > >>>>>> exten=s,n,Hangup() > >>>>>> > >>>>>> > >>>>>> And when I call 5001, the asterisk "craches" and in asterisk logs > show > >>>>>> the folow information: > >>>>>> > >>>>>> [Feb 23 18:15:22] WARNING[4142] app_transcoder.c: >Transcoding > >>>>>> [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,80008] > >>>>>> [Feb 23 18:15:22] ERROR[4142] app_transcoder.c: Error opening > encoder > >>>>>> [Feb 23 18:15:22] WARNING[4142] app_transcoder.c: -joining thread > >>>>>> > >>>>>> I receive rtsp streaming with mp4v video codec, and I want transcode > >>>>>> to H263 codec to softphone, the X-lite4. > >>>>>> > >>>>>> Any ideas??? > >>>>>> Help me please!!!! > >>>>>> > >>>>>> Best regards, > >>>>>> > >>>>>> Mário Dias > >>>>>> > >>>>>> -- > >>>>>> > _____________________________________________________________________ > >>>>>> -- 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 > >>>> > >>>> > >>>> > >>>> -- > >>>> _____________________________________________________________________ > >>>> -- 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 > -- Amit Anand +1 774 264-8024 +91 9013223047
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