similar to: Passing audio and video to linux

You could use RTMP protocol from asterisk to rtmpsrc in gstreamer. I used chan_rtmp long time back but it only supported audio back then. From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jerry Geis Sent: Friday, March 27, 2020 8:29 AM To: Development discussion of video media support in Asterisk <asterisk-video at lists.digium.com> Subject: Re:

I can create an audio call file and specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course), and I do specify also the Codecs: h264,h263 etc... Asterisk reports: *File /tmp/video does not

Hi all, I have a strange problem in my CentOS 6.4 (fully patched) workstation: I need to start pulseaudio maunally if I would like to use/listen sound ... And I don't understand why. These are a list of packages related to sound system installed in this workstation: alsa-lib-1.0.22-3.el6.x86_64 alsa-plugins-pulseaudio-1.0.21-3.el6.x86_64 alsa-utils-1.0.22-5.el6.x86_64

> -----Original Message----- > From: John Hodrien [mailto:J.H.Hodrien at leeds.ac.uk] > Sent: Friday, September 11, 2015 6:39 AM > To: CentOS mailing list > Subject: Re: [CentOS] C6 firefox esr h.264 support on youtube > > On Fri, 11 Sep 2015, Robert Arkiletian wrote: > > > does anyone running C6 have h.264 check box on www.youtube.com/html5 ? > > > > if

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On Fri, 11 Sep 2015, Denniston, Todd A CIV NAVSURFWARCENDIV Crane wrote: > If anyone has clues on how to build and add just a portion of > gst-plugins-bad into the existing EL OS set of gstreamer plugins, please > share (even URL pointers). > > [1] http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/openh264 > [2] https://en.wikipedia.org/wiki/OpenH264 > [3]

There are many Icecast source clients[1] available now. The Coolmic[2] app is particularly interesting because it can turn a Android phone (e.g. SmartLav+ lapel mic[3]) into a wearable mic for presenters. Not everybody is keen to run Icecast though, some people may want to mix-and-match the Icecast source clients like that with some other broadcasting platform, Jitsi Meet[4], Zoom or whatever.

Hello John, On Fri, 11 Sep 2015 14:18:50 +0100 (BST) John Hodrien <J.H.Hodrien at leeds.ac.uk> wrote: > On Fri, 11 Sep 2015, Denniston, Todd A CIV NAVSURFWARCENDIV Crane wrote: > > > If anyone has clues on how to build and add just a portion of > > gst-plugins-bad into the existing EL OS set of gstreamer plugins, please > > share (even URL pointers). > >

Hi all, I have a need to rebuild the gstreamer and gst-plugins in CentOS 7. So I found the wiki for rebuilding a source package... I downloaded the vault.centos.org files, installed them with dependencies. I did the rpmbuild -ba for each of the four packages. All seemed to go well. No errors reported. However - the /usr/bin/gst-inspect* listing shows the files as OLD ls -l /usr/bin/gst-inspect*

Thanks Jeremy. I am seeing questions similar to mine when I search - but I am not finding a solution. Was wondering if you know of something I have not found yet. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/92e1b497/attachment.html> -------------- next part -------------- --

On Fri, Sep 11, 2015 at 6:18 AM, John Hodrien <J.H.Hodrien at leeds.ac.uk> wrote: > On Fri, 11 Sep 2015, Denniston, Todd A CIV NAVSURFWARCENDIV Crane wrote: > > If anyone has clues on how to build and add just a portion of >> gst-plugins-bad into the existing EL OS set of gstreamer plugins, please >> share (even URL pointers). >> >> [1] >>

Does CentOS 8.1 support OLDER generate NVIDIA ? Like NVIDIA Corporation GF119M [GeForce GT 520M] I'm looking for hardware acceleration H264 type support. Thanks, Jerry

Hi - Is this list still alive ? How do I get audio/video into gstreamer from an asterisk call? Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/00b14154/attachment.html> -------------- next part -------------- --

Hello my hardware has: 03:00.0 VGA compatible controller: NVIDIA Corporation GT218 [ION] (rev a2) rpm -qa | grep nvidia nvidia-detect-410.78-1.el7.elrepo.x86_64 kmod-nvidia-340xx-340.107-2.el7_6.elrepo.x86_64 nvidia-x11-drv-340xx-340.107-1.el7.elrepo.x86_64 rpm -qa | grep libva libva-vdpau-driver-0.7.4-19.el7.x86_64 libva-utils-1.8.3-4.el7.x86_64 libva-devel-1.8.3-1.el7.x86_64

How do I tell from source rpm's: 1) the build order of gstreamer packages 2) the command line args for the ./configure For centos 7. Thanks, Jerry

Here's another problem I have with the 0.24 version of ffmpeg2theora. When I try to convert a h264 file to theora... (Note that for size and runtime reasons, foo.mts is a truncated file, I just took the first 32MB of the original file) ffmpeg2theora-0.24.linux32.bin foo.mts -x 1280 -y 720 -o foo-ffmpeg2theora-0.24.ogv Input #0, mpegts, from 'foo.mts': Duration: 00:00:15.83, start:

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open

Hello, This's my first ever post to CentOS lists. Sorry if I made any mistakes. My family is familiar with this RealPlayer Gold 11 and hesitate to use VLC or others. I googled a lot and tried different settings to no avail. Could you please tell me what I need to edit in order to slow down the audio/video playback of this old player? All other apps - even some older than Realplayer can play

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032