similar to: Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

I generate a groupedData object library(nlme) obj <- groupedData(mg10 ~ time | gp, data = common, outer = ~pct) gp has 101 levels, and pct has 3. There are 38, 25, 38 gps in each of the levels of pct respectively. I fit my model fit.rtg <- lme(mg10 ~ time * group, data = obj, random = ~time * group | gp) Now I try to plot the results. I would like to print 40 panels on each

Hi Folks I recently got back the message below form a company I suggested supporting WINE with their software. Is there any answer - or is it, as they suggest, a lost cause? The error I get from WINE is: RTG Bills (title) ADO error 1AD ActiveX component can't create object [OK](button) Is there anyway to have ActiveX & ADO work with WINE? Regards, Morgan. > No, RTG Bills won't

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

Hello RoR folks, I am using Instant Rails and a small app that currently uses scaffolding for CRUD on just one table with 7 attributes. I am not shure wether scaffolding slows down the app that much but I have a time lag for every request with above 4 sec. The preinstalled apps like Type are running damn fast, which is not the case for my stuff. Any help on this would be great Thanks, RTG --

Hi, I am applying the following code to map pca loadings onto a 3d grid, my problem is this - the output only plots the spheres in the requested color (in this case "red") for the first argument. The sphere from the second argument appear as flat dark circles. Also the text3d() command only seems to work for a couple of the positions, with no text added in most cases. Could anyone offer

Hi! I tried this in features.conf testfeature => *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T and it works... but... I would be able to transfer a call to any phone number, so I tried to use this line: testfeature => _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T but... Asterisk crash! (it doesn't want even to reload configuration) Any idea about how to do so? Thanks a lot!

Andrew Haley brought up this interesting issue on the GCC mailing-list that directly affect the stability of the ARM llvm target when using multi-threading. http://gcc.gnu.org/ml/gcc-patches/2009-08/msg00600.html basically using __sync_synchronize() with GCC on ARM does not generate any code for the fence. For what I know: The only working fix for this issue on Linux would be to create a call to

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

Hi, I have a asterisk/voip newbie and I am sorry if my quetion is banal. I used in my private LAN, Express Talk on Windows XP and Asterisk latest version on Fedora Core 4 , with this configuration in Express Talk Lines menu: Setting for Line: Default Line Settings Full 'friendly' Display Name: port SIP Numeber: 200 Server: 10.0.0.112 Password: mypassword In menu Network: Local SIP Port

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

Hi all, I'm using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

https://bugzilla.samba.org/show_bug.cgi?id=6276 Summary: crtimes.patch does not preserve creation dates on Mac x86_64 only Product: rsync Version: 3.0.6 Platform: x86 OS/Version: Mac OS X Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi All I downloaded the binary package of Samba 2.2.2 on Friday from one of the Samba mirror sites, I think Belgium, and installed it on a Sun E450 running Solaris 2.6. Today everything has fallen flat on its face. Unfortunately I am on holiday this week so not able to deal with the problem directly, but via telephone calls. Apparently what is happening is that smbd is gobbling up all the

Hello everyone, Btrfs v0.14 is now available for download. Please note the disk format has changed, and it is not compatible with older versions of Btrfs. For downloads and documention, please see the Btrfs project page: http://btrfs.wiki.kernel.org The oss.oracle.com Btrfs project page will soon start redirecting here. v0.14 has a few performance fixes and closes some races that could have

Dear members of the list; I am writing in the hope to get some help with a very peculiar problem with my new asterisk 1.6.0.1 installation. The same code runs on version 1.2 without problems, but it seems the behaviour has changed (also on 1.4.7, which I tried). Please consider the following extension: exten => 1213,1,Answer exten => 1213,n,Set(counter=X) exten =>

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

I have a bunch of sol10U8 boxes with ZFS pools, most all raidz2 8-disk stripe. They''re all supermicro-based with retail LSI cards. I''ve noticed a tendency for things to go a little bonkers during the weekly scrub (they all scrub over the weekend), and that''s when I''ll lose a disk here and there. OK, fine, that''s sort of the point, and they''re

================================================================= "I don't know anything about music. In my line you don't have to." Elvis Presley ================================================================= Release Announcements ===================== This is the latest stable release of Samba 3.4. Major enhancements in Samba 3.4.1 include:

================================================================= "I don't know anything about music. In my line you don't have to." Elvis Presley ================================================================= Release Announcements ===================== This is the latest stable release of Samba 3.4. Major enhancements in Samba 3.4.1 include: