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List: asterisk-video
Subject: Re: [Asterisk-video] Video with asterisk,
From: andrew mmc <andrewmmc () gmail ! com>
Date: 2010-02-09 12:25:39
Message-ID: ed4c5aa71002090425m63198a1cy1e9ff25dc29cdc32 () mail ! gmail ! com
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Hello,
I'm going through the same thing. Perhaps we can cooperate - it would be
nice to get a nice open-source HowTo to get this working - it is quite a
challenge piecing everything together. However, progress is being made and
actually on the face of it, it isn't that hard.
It isn't a problem with 1.6...
The problem you are having there is that the swscale library isn't being
linked in. I had this problem and fixed it but it was long enough now that
I have forgotten exactly what I did.
But basically, you need to ensure that ffmpeg is built properly, including
libavcodec and libswscale, and you are linking all the right libraries with
app_transcoder. i.e. in the Makefile you have -lpthread -lavcodec -lswscale
as your library flags when linking.
When it is right it kicks up a few warnings, but compiles ok.
Andrew
On Tue, Feb 9, 2010 at 3:16 AM, Pham Quy <quyps@vega.com.vn> wrote:
> Hi,
>
> Thank to all your kind reply,
>
> I downloaded this packages
> http://asteriskvideo.svn.sourceforge.net/viewvc/asteriskvideo/
>
> and followed this guide
>
> http://asterisk-party.net/index.php/Asterisk_Video_3G_FR
>
> As i understand what i do need is that: installing H324 and AMR in order
> to make 3G calls, app_mp4 to playback mp4 video, app_rtsp for video
> streaming, app_transcoder to use my webcam.
>
> Since, I just need to make a video IVVR, or video call with SIP phones,
> i just need to buid app_mp4 (to playback), app_rtsp, and app_transcoder,
> is that right?
>
> To make a video call, I successfully build app_rtsp and load into
> Asterisk (1.6) but still have some problem with app_transcoder the
> message from asterisk as following:
>
> --------------------------------------
> [Feb 9 09:08:23] WARNING[4672]: loader.c:428 load_dynamic_module: Error
> loading
> 'app_transcoder.so': /usr/lib/asterisk/modules/app_transcoder.so:
> undefined symbol: sws_scale
> [Feb 9 09:08:23] WARNING[4672]: loader.c:781 load_resource: Module
> 'app_transcoder.so' could not be loaded.
> --------------------------------------
>
> I guess it due to incompatibility of asterisk 1.6, how can I overcome
> this problem?
>
> There is a simple dialplan in the guide
>
> ---------------------------------------------
> [default]
> exten => 5003.1, Answer ()
> exten => 5003, n, RTSP (rtsp: / / 192.168.1.1/live.sdp)
> exten => 5003, n, Hangup ()
> ---------------------------------------------
>
> I dont really understand the call RTSP(rtsp://192.168.1.1/live.sdp)
> what argument should i pass in RTSP()? What is the IP?
>
> Did i ask to much!!? ;)
>
> plz help.
> Quyps
>
>
> On Mon, 2010-02-08 at 10:49 +0100, Borja SIXTO wrote:
> > That's true,
> >
> > But if want to play video menu, you need to use a video converter tool.
> > i6net provide a tool called mp4asterisk (for i6net's video users only
> > for the moment).
> >
> > It generates a "standard" Asterisk .h263, h263p or .h264 file from a 3GP
> > or MP4 file (we provide a mp4tool to generate mp4 files from and avi
> > video file).
> > This allows you to use the playback application to play a simple video.
> >
> > The mp4play is more powerful because you can add in the same clip
> > different tracks to be able to play the video in different codecs
> profiles.
> >
> > Le 08/02/2010 10:09, Jesús Gumiel a écrit :
> > > Hi Quy,
> > >
> > > one simple option to create your basic IVVR is record your own videos
> > > with the command record in Asterisk. If you have your peer configured
> > > with the option videosupport=yes, you can do a call to the dialplan
> > > and record a message.
> > >
> > > The command "Record" create two files, one with the audio y other with
> > > the video. For that you must configure the codecs correctly in your
> > > peer. After recording you could use the command "Playback" to see the
> > > video on your phone.
> > >
> > > This is valid with a VoIP device, if you want use 3G call, you would
> > > need to configure the h324m stack like Sixto said.
> > >
> > > 2010/2/4 Pham Quy<quyps@vega.com.vn>:
> > >
> > >> hi!
> > >>
> > >> I'm new with *. I need to make a demostration showing a IVVR on
> asterisk. I
> > >> found some information here but it seem like its cost (Vxi, or
> Diastar) are
> > >> quite expensive.
> > >>
> > >> Is there a way i can build an simple video response system without
> that
> > >> expense?
> > >>
> > >> Thanks.
> > >>
> > >> Quyps
> > >> --
> > >> _____________________________________________________________________
> > >> -- 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
> > >>
> > >>
> > >
> >
> > --
> > Borja Sixto, Research& Innovation - http://www.i6net.com
> > Office: +34 911877477 | Gtalk: bsixto@gmail.com | Skype: borja.sixto
> >
> >
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
[Attachment #5 (text/html)]
Hello,<div><br></div><div>I'm going through the same thing. Perhaps we can \
cooperate - it would be nice to get a nice open-source HowTo to get this working - it \
is quite a challenge piecing everything together. However, progress is being made \
and actually on the face of it, it isn't that hard.</div> \
<div><br></div><div><div>It isn't a problem with 1.6...</div><div> </div><div>The \
problem you are having there is that the swscale library isn't being linked in. \
I had this problem and fixed it but it was long enough now that I have forgotten \
exactly what I did.</div> </div><div><br></div><div>But basically, you need to ensure \
that ffmpeg is built properly, including libavcodec and libswscale, and you are \
linking all the right libraries with app_transcoder. i.e. in the Makefile you have \
-lpthread -lavcodec -lswscale as your library flags when linking.</div> \
<div><br></div><div>When it is right it kicks up a few warnings, but compiles \
ok.<br><br></div><div>Andrew</div><div><br><div class="gmail_quote">On Tue, Feb 9, \
2010 at 3:16 AM, Pham Quy <span dir="ltr"><<a \
href="mailto:quyps@vega.com.vn">quyps@vega.com.vn</a>></span> wrote:<br> \
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc \
solid;padding-left:1ex;">Hi,<br> <br>
Thank to all your kind reply,<br>
<br>
I downloaded this packages<br>
<a href="http://asteriskvideo.svn.sourceforge.net/viewvc/asteriskvideo/" \
target="_blank">http://asteriskvideo.svn.sourceforge.net/viewvc/asteriskvideo/</a><br>
<br>
and followed this guide<br>
<br>
<a href="http://asterisk-party.net/index.php/Asterisk_Video_3G_FR" \
target="_blank">http://asterisk-party.net/index.php/Asterisk_Video_3G_FR</a><br> <br>
As i understand what i do need is that: installing H324 and AMR in order<br>
to make 3G calls, app_mp4 to playback mp4 video, app_rtsp for video<br>
streaming, app_transcoder to use my webcam.<br>
<br>
Since, I just need to make a video IVVR, or video call with SIP phones,<br>
i just need to buid app_mp4 (to playback), app_rtsp, and app_transcoder,<br>
is that right?<br>
<br>
To make a video call, I successfully build app_rtsp and load into<br>
Asterisk (1.6) but still have some problem with app_transcoder the<br>
message from asterisk as following:<br>
<br>
--------------------------------------<br>
[Feb 9 09:08:23] WARNING[4672]: loader.c:428 load_dynamic_module: Error<br>
loading<br>
'app_transcoder.so': /usr/lib/asterisk/modules/app_transcoder.so:<br>
undefined symbol: sws_scale<br>
[Feb 9 09:08:23] WARNING[4672]: loader.c:781 load_resource: Module<br>
'app_transcoder.so' could not be loaded.<br>
--------------------------------------<br>
<br>
I guess it due to incompatibility of asterisk 1.6, how can I overcome<br>
this problem?<br>
<br>
There is a simple dialplan in the guide<br>
<br>
---------------------------------------------<br>
[default]<br>
exten => 5003.1, Answer ()<br>
exten => 5003, n, RTSP (rtsp: / / <a href="http://192.168.1.1/live.sdp" \
target="_blank">192.168.1.1/live.sdp</a>)<br> exten => 5003, n, Hangup ()<br>
---------------------------------------------<br>
<br>
I dont really understand the call RTSP(rtsp://<a href="http://192.168.1.1/live.sdp" \
target="_blank">192.168.1.1/live.sdp</a>)<br> what argument should i pass in RTSP()? \
What is the IP?<br> <br>
Did i ask to much!!? ;)<br>
<br>
plz help.<br>
Quyps<br>
<div><div></div><div class="h5"><br>
<br>
On Mon, 2010-02-08 at 10:49 +0100, Borja SIXTO wrote:<br>
> That's true,<br>
><br>
> But if want to play video menu, you need to use a video converter tool.<br>
> i6net provide a tool called mp4asterisk (for i6net's video users only<br>
> for the moment).<br>
><br>
> It generates a "standard" Asterisk .h263, h263p or .h264 file from a \
3GP<br> > or MP4 file (we provide a mp4tool to generate mp4 files from and avi<br>
> video file).<br>
> This allows you to use the playback application to play a simple video.<br>
><br>
> The mp4play is more powerful because you can add in the same clip<br>
> different tracks to be able to play the video in different codecs profiles.<br>
><br>
> Le 08/02/2010 10:09, Jesús Gumiel a écrit :<br>
> > Hi Quy,<br>
> ><br>
> > one simple option to create your basic IVVR is record your own videos<br>
> > with the command record in Asterisk. If you have your peer configured<br>
> > with the option videosupport=yes, you can do a call to the dialplan<br>
> > and record a message.<br>
> ><br>
> > The command "Record" create two files, one with the audio y other \
with<br> > > the video. For that you must configure the codecs correctly in \
your<br> > > peer. After recording you could use the command \
"Playback" to see the<br> > > video on your phone.<br>
> ><br>
> > This is valid with a VoIP device, if you want use 3G call, you would<br>
> > need to configure the h324m stack like Sixto said.<br>
> ><br>
> > 2010/2/4 Pham Quy<<a \
href="mailto:quyps@vega.com.vn">quyps@vega.com.vn</a>>:<br> > ><br>
> >> hi!<br>
> >><br>
> >> I'm new with *. I need to make a demostration showing a IVVR on \
asterisk. I<br> > >> found some information here but it seem like its cost \
(Vxi, or Diastar) are<br> > >> quite expensive.<br>
> >><br>
> >> Is there a way i can build an simple video response system without \
that<br> > >> expense?<br>
> >><br>
> >> Thanks.<br>
> >><br>
> >> Quyps<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>
> Borja Sixto, Research& Innovation - <a href="http://www.i6net.com" \
target="_blank">http://www.i6net.com</a><br> > Office: +34 911877477 | Gtalk: <a \
href="mailto:bsixto@gmail.com">bsixto@gmail.com</a> | Skype: borja.sixto<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></div></div></blockquote></div><br></div>
--
_____________________________________________________________________
-- 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
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