humm.. are you shure??<div><br></div><div>Well, in total, what dependences, or codecs, or apps, that I have to install??</div><div><br></div><div>AMR codec (will install), app_transcoder (installed), app_rtsp (installed for streaming), ffmpeg (installed) and more???</div>
<div><br></div><div>Best Regards,</div><div><br></div><div>Mário Dias</div><div><br><br><div class="gmail_quote">2011/2/25 amit anand <span dir="ltr"><<a href="mailto:onewaytoconnect@gmail.com">onewaytoconnect@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi<br><br>this is due to codec amr is not properly installed<div><div></div><div class="h5"><br><br><div class="gmail_quote">
On Fri, Feb 25, 2011 at 6:06 PM, Mário Dias <span dir="ltr"><<a href="mailto:mario@hardserver.com" target="_blank">mario@hardserver.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Hello agian!<br>
<br>
I forgot another error in asterisk logs:<br>
<br>
[Feb 25 18:03:30] WARNING[18705] app_transcoder.c: >Transcoding<br>
<div>[,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,180004]<br>
</div>[Feb 25 18:03:30] WARNING[18707] app_rtsp.c: >rtsp play<br>
[Feb 25 18:03:31] WARNING[18707] channel.c: Unable to find a codec<br>
<div>translation path from 0x780004 (ulaw|h263|h263p|h264) to 0x2000<br>
(nothing)<br>
</div>[Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received<br>
from '192.168.0.89'<br>
[Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received<br>
from '192.168.0.89'<br>
[Feb 25 18:03:31] NOTICE[18705] rtp.c: Unknown RTP codec 126 received<br>
from '192.168.0.89'<br>
<br>
What is the problem????<br>
<div><div></div><div><br>
2011/2/25 Mário Dias <<a href="mailto:mario@hardserver.com" target="_blank">mario@hardserver.com</a>>:<br>
> Hello! I just try reinstall ffmpeg in other version of linux (ubuntu)<br>
> and the before error not appear now.<br>
><br>
> But, When I call 5001, the video call answer but not appear the video<br>
> (waitting remote video) in X-lite4.<br>
><br>
> In asterisk logs there are:<br>
><br>
> [Feb 25 17:46:54] WARNING[18490] app_transcoder.c: >Transcoding<br>
> [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,180004]<br>
> [Feb 25 17:46:54] WARNING[18492] app_rtsp.c: >rtsp play<br>
> [Feb 25 17:46:54] WARNING[18492] channel.c: Unable to find a codec<br>
> translation path from 0x780004 (ulaw|h263|h263p|h264) to 0x2000<br>
> (nothing)<br>
><br>
> why ???<br>
><br>
> I remember that I allowed in sip.conf : video support, h263, h263p, h264<br>
><br>
> I want transcode the codec of received video RTSP streaming (codec<br>
> mp4v) to H263 of my softphone.....<br>
><br>
><br>
><br>
><br>
> 2011/2/25 Mário Dias <<a href="mailto:mario@hardserver.com" target="_blank">mario@hardserver.com</a>>:<br>
>> Sergio,<br>
>><br>
>> The results of command ffmpeg -formats | grep h263<br>
>><br>
>><br>
>> asterisk2:/# ffmpeg -formats | grep h263<br>
>> FFmpeg version r11872+debian_0.svn20080206-18+lenny3, Copyright (c)<br>
>> 2000-2008 Fabrice Bellard, et al.<br>
>> configuration: --enable-gpl --enable-libfaad --enable-pp<br>
>> --enable-swscaler --enable-x11grab --prefix=/usr --enable-libgsm<br>
>> --enable-libtheora --enable-libvorbis --enable-pthreads<br>
>> --disable-strip --enable-libdc1394 --disable-armv5te --disable-armv6<br>
>> --disable-altivec --disable-vis --enable-shared --disable-static<br>
>> libavutil version: 49.6.0<br>
>> libavcodec version: 51.50.0<br>
>> libavformat version: 52.7.0<br>
>> libavdevice version: 52.0.0<br>
>> built on Feb 13 2011 03:56:05, gcc: 4.3.2<br>
>> DE h263 raw h263<br>
>> D VSDT h263<br>
>> D VSD h263i<br>
>> even though both encoding and decoding are supported. For example, the h263<br>
>> decoder corresponds to the h263 and h263p encoders, for file formats it is even<br>
>><br>
>><br>
>> and now?? What I have to do to solve my issue??<br>
>><br>
>> Best regards,<br>
>><br>
>> Mário Dias<br>
>><br>
>><br>
>>> 2011/2/24 Sergio Garcia Murillo <<a href="mailto:sergio.garcia@fontventa.com" target="_blank">sergio.garcia@fontventa.com</a>>:<br>
>>>> The app_transcoder is loaded correctly:<br>
>>>><br>
>>>> [Feb 23 18:15:22] ERROR[4142] app_transcoder.c: Error opening encoder<br>
>>>><br>
>>>> Could you check if your libavcodec.so library supports h263 encoding?<br>
>>>><br>
>>>>>ffmpeg -formats | grep h263<br>
>>>> DE h263 raw H.263<br>
>>>><br>
>>>> BR<br>
>>>> Sergio<br>
>>>><br>
>>>> El 24/02/2011 21:51, Mitul Limbani escribió:<br>
>>>>><br>
>>>>> Hi Mario,<br>
>>>>><br>
>>>>> Can you check if the app_transcoder.so got loaded without any problem<br>
>>>>> within Asterisk Startup ?<br>
>>>>><br>
>>>>> you can try this:<br>
>>>>><br>
>>>>> core set verbose 5<br>
>>>>> module unload app_transcode.so<br>
>>>>> module load app_transcode.so<br>
>>>>><br>
>>>>> and paste the output.<br>
>>>>><br>
>>>>> Regards,<br>
>>>>> Mitul Limbani<br>
>>>>> Enterux Solutions,<br>
>>>>> <a href="http://www.enterux.com" target="_blank">www.enterux.com</a><br>
>>>>><br>
>>>>> Quoting Mário Dias <<a href="mailto:mario@hardserver.com" target="_blank">mario@hardserver.com</a>>:<br>
>>>>><br>
>>>>>> Hello! I just installed the app_transcoder with success and this runs<br>
>>>>>> well with asterisk boot...<br>
>>>>>><br>
>>>>>> Now the problem is:<br>
>>>>>><br>
>>>>>> My extensions.conf:<br>
>>>>>><br>
>>>>>> [default]<br>
>>>>>><br>
>>>>>> exten=5001,1,Answer()<br>
>>>>>><br>
>>>>>> exten=5001,n,Transcode(,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50)<br>
>>>>>> exten=5001,n,Hangup()<br>
>>>>>><br>
>>>>>> [camera]<br>
>>>>>><br>
>>>>>> exten=s,1,Answer()<br>
>>>>>> exten=s,n,Rtsp(rtsp://<a href="http://192.168.10.14:8554/CH001.sdp" target="_blank">192.168.10.14:8554/CH001.sdp</a>)<br>
>>>>>> exten=s,n,Hangup()<br>
>>>>>><br>
>>>>>><br>
>>>>>> And when I call 5001, the asterisk "craches" and in asterisk logs show<br>
>>>>>> the folow information:<br>
>>>>>><br>
>>>>>> [Feb 23 18:15:22] WARNING[4142] app_transcoder.c: >Transcoding<br>
>>>>>> [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50,80008]<br>
>>>>>> [Feb 23 18:15:22] ERROR[4142] app_transcoder.c: Error opening encoder<br>
>>>>>> [Feb 23 18:15:22] WARNING[4142] app_transcoder.c: -joining thread<br>
>>>>>><br>
>>>>>> I receive rtsp streaming with mp4v video codec, and I want transcode<br>
>>>>>> to H263 codec to softphone, the X-lite4.<br>
>>>>>><br>
>>>>>> Any ideas???<br>
>>>>>> Help me please!!!!<br>
>>>>>><br>
>>>>>> Best regards,<br>
>>>>>><br>
>>>>>> Mário Dias<br>
>>>>>><br>
>>>>>> --<br>
>>>>>> _____________________________________________________________________<br>
>>>>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
>>>>>><br>
>>>>>> asterisk-video mailing list<br>
>>>>>> To UNSUBSCRIBE or update options visit:<br>
>>>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
>>>>>><br>
>>>>><br>
>>>>><br>
>>>>> --<br>
>>>>> _____________________________________________________________________<br>
>>>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
>>>>><br>
>>>>> asterisk-video mailing list<br>
>>>>> To UNSUBSCRIBE or update options visit:<br>
>>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
>>>><br>
>>>><br>
>>>><br>
>>>> --<br>
>>>> _____________________________________________________________________<br>
>>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
>>>><br>
>>>> asterisk-video mailing list<br>
>>>> To UNSUBSCRIBE or update options visit:<br>
>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
>>>><br>
>>><br>
>><br>
><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div></div><div><br></div></div></div><div>Amit Anand</div><div><br></div><div><span style="font-family:arial,sans-serif;font-size:medium;border-collapse:collapse"><h1 style="margin:12px 5px 5px 10px;padding:0px;font-family:arial,sans-serif;font-size:16px;color:rgb(0, 0, 0)">
<span style="padding-right:10px">+1 774 264-8024</span></h1></span></div><div>+91 9013223047</div><div><img></div><br>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br></blockquote></div><br></div>