similar to: Converting asterisk h264 recordings

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some ?official? how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial("SIP/dev02-08c36f28", "SIP/3246 at 989800-out||W") in new stack Jan 23 09:02:07 DEBUG[2236]

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

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

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

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 asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

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

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 ->

https://bugs.freedesktop.org/show_bug.cgi?id=86006 Bug ID: 86006 Summary: [NV84] Nvidia GeForce 8600 GT VDPAU h264 hardware acceleration Product: xorg Version: 7.6 (2010.12) Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

Hey, all. I'm currently trying out GlusterFS 3.3. I've got two servers and four clients, all on separate boxes. I've got a Distributed-Replicated volume with 4 bricks, two from each server, and I'm using the FUSE client. I was trying out failover, currently testing for reads. I was reading a big file, using iftop to see which server was actually being read from. I put up an

Hello list, what is the difference between these 2 codecs ? What codec to choose if bandwith is an issue ? (like in most cases I guess) Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101208/d611534a/attachment.htm

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

I've made packages for Debian Squeeze for Gluster 3.4.0-alpha2, they are available on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/. They built and installed successfully, and have been running nicely for a couple of hours, but your mileage may vary. The Debian packaging is on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/glusterfs-3.4.0-debian.tar.gz. I took the

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen

Dear list, I'm looking for a function to generate (simulate) a random Weibull point process. Can anyone help? Cheers, Torbj?rn Ergon, University of Oslo