similar to: Compiz Key binding and application key bindings

What is the minimum version required for Compiz to run? Is it OpenGL 1.5 or OpenGL 2.0? -Bipin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freedesktop.org/archives/compiz/attachments/20090126/985c425a/attachment.htm

Dear Manoj, Have found out the mistake in my script. One must create directory if he/she wants to move old files to backup directory. In my script I was not creating any new backup-dir(datewise), therefore despite of have -b --backup-dir flags it was not working. This means rsync can only create automatically new direcotry for destination but not backup-dir !!!! Am I correct? Bipin ---

Hello, I have Novell Netware File server. Which is mounted as /mnt/novell on one of my Linux m/c. I want to take backup of Novell Server to my Linux m/c. I need everyday backup of Novell server to be taken on Linux m/c. I don't want full backup every day but I need an incremental backup. I do not want to delete any old directory or files. I have taken script from rsync examples, and

Hello all, I am facing as serious problem when running asterisk in HP server.We are developing application to make the outbound calls in PRI lines .We normally uses IBM machine as our servers ,and it was working fine for all installation.For the cost reduction we this time tried with HP server. Model(HP proliant ml110). When we make the calls the there is a lots of disturbance in the sound even

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

I'm hitting an issue where legalizeOp is overflowing the stack. Are there any recommended ways of getting around this? The bitcode that causes this issue is attached. Thanks, Micah Villmow Systems Engineer Advanced Technology & Performance Advanced Micro Devices Inc. S1-609 One AMD Place Sunnyvale, CA. 94085 P: 408-749-3966 -------------- next part -------------- An

Greetings! We are using cisco 7940 phone with SIP and asterisk. We would like to be able to have phone directories available on the phones that are sourced from active directory. Are their any scripts that can connect to the AD server via LDAP and then create the directory.html file for the phones? Thanks! Liz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings listers, I am running Asterisk 1.4.21.2 on Suse 11.0 on a Dual Processor Dell Poweredge 1650. I recently attempted to update the BIOS and now have this happen: When the machine starts up, Asterisk runs fine. When I do a large wget or scp, the local SIP to SIP quality goes to heck in a handbasket. The only resolution I've found so far is to completely

Hello, I'am following the beginning tutorial on the Wiki, but actually, I can't even start the server correctly so far. The imap-login child dies immediately. What I can guess from the log is that it tries some unpermietted operations on some files but I can't find out which files are concerned. Google is not my friend at all either on this matter. How can I know which files are

Hi, It seems BRI signalling settings are missing from http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf I would like to add those parameters : bri_cpe_ptmp bri_cpe bri_net Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conftied to a specific Asterisk version ? Can I edit this ? Regards -------------- next part -------------- An HTML attachment was

Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with

Hi, The attached patch extends the mux listener to accept SOCKS requests in addition to the native mux commands. The rationale behind is that creating tunnels attached to TCP ports is a security hazard in multi-user machines where there is no way to control who connects through the tunnels. On the other hand, The mux UNIX domain socket binds to the file system and regular permissions can be

I''m working on a mapstraction implementation modeled after the radius search feature. (http://mapstraction.com/demo-radius.php) I''d like to recognize when the user extends the radius beyond the visible edges of the map and adjust the zoom level. I''m not seeing how to test this when redrawing the circle. I thought maybe I wanted to use polylineCenterAndZoom()

Google Summer of Code has announced the opening of applications for Mentor organizations. Pamela Fox mentioned awhile ago that Mapstraction may be a good fit. http://socghop.appspot.com/ I would like to brainstorm if we have identifiable projects, and preferably even interested students, that would like to work together on putting in some applications. A couple of initial possibilities: -

Hi, I am trying to plot the Case-Shiller index found at: http://www2.standardandpoors.com/spf/pdf/index/CSHomePrice_History_022445.xls The way I'm importing it into R is as follows: library(gdata) W <- read.xls("http://www2.standardandpoors.com/spf/pdf/index/CSHomePrice_History_022445.xls ", header=TRUE) attach(W) To give you and idea of what the data looks like: >

I am about to setup a new machine and based on a thread in the freetel-oslec list, I came across the idea of compiling Intel optimizations in when using oslec w/ dahdi. So I edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_config.h to #define CONFIG_DAHDI_MMX which on its own wont compile. It was suggested to edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/Kbuild

Hi, My setup is: IPPhone1 --- Asterisk1 with B410P ---- Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx "pri show spans" keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned,

Problems building celt-0.5.2 for blackfin. Use the following to configure: #!/bin/sh # bfinconf # Set up configuration to cross compile on blackfin ./configure \ --host=bfin \ --enable-fixed-point \ CC=bfin-uclinux-gcc \ CFLAGS='-O2 '\ '-mfast-fp '\ '-ffast-math '\ '-Wall -Dlinux '\ '-D__linux__ '\ '-fno-strict-aliasing '\

Came across this today http://www.techcrunch.com/2009/03/09/y-combinators-voxli-targets-gamers- with-browser-based-group-voice-chat/ Yet another opportunity Asterisk <http://www.digium.com/> / Mexuar <http://www.mexuar.com/> / PhoneFromHere <http://www.phonefromhere.com/> / (insert the other multiple versions here) have missed out on. Lol - and not from lack of

Hi all! On outgoing call sometimes Asterisk use/give back the caller id sent back by called number instead of number called by me. This is annoying and misleading statistics if other side use some exotic number. For example I have called number 12345678 and CDR include the number 333 as callerid which was sent back by called number/set/switch/whatever. Normally it cannot be an issue but I have