similar to: Video codecs: H263 & H264

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

Hello everyone, We are using an Asterisk server (Ubuntu 10.04 64 bits, Asterisk 1.6.2.7, TDM2400 - 12 FXS/12 FXO), but I noticed the following dmesg alert: asterisk[4687]: segfault at 60 ip ****** sp ****** error 4 in asterisk[400000+187000] The server is running without troubles. I know I should worry, just don't know where to start. Any advices? Alex Saavedra -------------- next part

Here is something that could change a lot of thinks on video/ audio streaming on the web http://www.kaourantin.net/2007/08/what-just-happened-to-video-on-web_20.html a shame they did not add theora/vorbis. -- %<------------------------------------------------------->% Michel memeteau VOIP | Visio: sip:freechelmi@gizmoproject.com Fixe : 0491886375 !!! APARTIR du 15/09/2007 ->

Here is something that could change a lot of thinks on video/ audio streaming on the web http://www.kaourantin.net/2007/08/what-just-happened-to-video-on-web_20.html a shame they did not add theora/vorbis. -- %<------------------------------------------------------->% Michel memeteau VOIP | Visio: sip:freechelmi@gizmoproject.com Fixe : 0491886375 !!! APARTIR du 15/09/2007 ->

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

Hello, We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

By browsing on the mailing list I learned that its possible to generate .h263 asterisk friendly files with gstreamer. The script below it's supposed to do just that, however I get error when trying it out locally. gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux ! ffdec_h263 ! videoscale ! video/x-raw-yuv,width=352,height=288 ! ffenc_h263 rtp-payload-size=512 !

I'm trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP). Problem is my video system only speaks H263-2000 version (aka H263++). As far as I can see, * only understands H263 and H263+ in and sdp. Can anybody tell me how to extend asterisk so it'll support H263++?

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

Buenas noches, Soy nueva en R y a veces me cuesta pensar los cálculos de manera más práctica, por los que les agradecería la ayuda. Tengo una matriz de datos con una dim = 35745 19, correspondientes a ubicaciones de 39 perros, cada perro tiene poco más o poco menos de 1000 datos. Necesito saber el % de uso de hábitat natural. Es un sencillo cálculo de porcentaje, pero me gustaría hacerlo mas

Hola, estoy tratando de hacer un boxplot para compara diferentes comportamientos. Tengo un set de datos con 17 columnas, quiero crear un boxplot considerando 3 de ellas. He buscado en foros y siempre responden esta pregunta con subconjuntos de datos dentro de una misma columna, y no he logrado realizarlo. Este es un ejemplo de mis datos: ID: id animal EP: episodio AA: acostado y alerta AD:

Hi, I'm using the Record dialplan Application in an Context. My goal is to get a single screenshot of the h264 media stream per call. same => n,Record(/tmp/test.wav,0,10,qk) I nicely get a File test.h264. Is there a way to Playback this h264 video file on my computer or convert it somehow? VLC can't take it somehow. Regards, Benjamin -------------- next part -------------- An HTML

Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up.

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

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:

Hi all, I have to compare the theora codec with MPEG and H264. I was googling and I found that the PSNR is a common used parameter. How can I do this with Theora? Thanks -- Leonardo de Paula Rosa Piga Undergraduate Computer Engineering Student LSC - IC - UNICAMP http://www.students.ic.unicamp.br/~ra033956

https://bugs.freedesktop.org/show_bug.cgi?id=93573 Bug ID: 93573 Summary: bad performance with 4k h264 video Product: xorg Version: git Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee: nouveau at lists.freedesktop.org