similar to: bug: copying unreadable file from remote host returns errorlevel 0

Hello, http://rsync.samba.org/cvs.html gives 404 Not Found. I do not know if it is ok or not ;) Thanks, --alexm

Hi -I am using R v 2.13.0. I am trying to use the aggregate function to calculate the percent at length for each Trip_id and CommonName. Here is a small subset of the data. Trip_id Vessel CommonName Length Count 1 230 Sunlight Shad,American 19 1 2 230 Sunlight Shad,American 20 1 3 230 Sunlight Shad,American 21

Hello While searching the archives and googling for : -login script to map drives according to group membership I saw lots of complicated solutions (on-the-fly scripts, group directories) etc. but the following works very well for me: I downloade dthe ifmember.exe from the microsoft website, and stuck it in the netlogon directory (not the scripts directory)

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was

Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi

Dear All, I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not be able to upgrade to release 1 until the patches catch up and I have done the required testing. The system was very stable until two days ago. The changes made

Full_Name: Russell V. Lenth Version: 2.3.1 OS: Windows XP Pro Submission from: (NULL) (128.255.132.188) I wrote a simple .BAT file to run the Sweave function on a file (via RTerm), then run pdflatex on the result (after RTerm exits). The issue is that if an error condition occurs in RTerm, it is prudent to not do the pdflatex processing afterward. Here are the relevant statements in the BAT

I am unable to connect to build master at:google1.osuosl.org:9990 from my buildslave. See attached twisted9.0.0 log. 2010-02-12 09:50:03-0800 [-] Log opened. 2010-02-12 09:50:03-0800 [-] twistd 9.0.0 (C:\Python26\python.exe 2.6.2) starting up. 2010-02-12 09:50:03-0800 [-] reactor class: twisted.internet.win32eventreactor.Win32Reactor. 2010-02-12 09:50:03-0800 [-] Starting factory

Dear Group, I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi

Kerry's post looks very promising: | It can but is probably not as nice as you would like. It's a huge improvement over what I can do now. Thank you! | flac -t *.flac Now, that doesn't work for me. I know that command.com doesn't glob wildcards but rather expects the application to do it, and apparently flac.exe doesn't. Under a real shell that wouldn't be a problem,

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Would this work? The problem is, I do not have roaming profiles in use(As it is not practical for us, as users tend to use the same computers). I want only users with IP from 192.168.1.x to load the databases. The loaddatabases.cmd file in question would contain rem Grab IP address and dump to a file ipconfig >"%userprofile%\ipconfig.txt" echo Checking for your location

Dear Group, I have been successful in setting up the Agents, queues and getting agents to log in. Is there a way that I could configure the system so that the agent is called back. i.e. the agent logs into the system, a call is destined for them and their phone rings. If some one has this setup I would be very interested in hearing from them. Warm Regards and Thanks --------------- Shad

I''m using ferret (0.9.4) in rails, but outside of the "acts_as_ferret" plugin. Whenever I use a QueryFilter (even a very simple one), the server will crash after one, two, or three reloads of a page (same page, same query, same filter). It''s very non-deterministic and I can''t seem to reproduce it outside of my application environment (I can''t get it

Back on Oct 31st Karanbir said he would take care of the missing centos-release srpm. It still hasn't shown up on any of the mirrors. Is this ever going to be fixed? -Shad

Dear All, I have a number of call queues defined in Asterisk. I would like to program a system attendant that tells people; 1. Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly

Dear Group, At the moment I use SJPhone as my soft phone with Asterisk. I prefer the look and feel of the x-ten lite. However, when ever I use my x-ten lite I get a lot of breakup in my communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I don't have the same issue with SJPhone. I'm sure this is a configuration issues, but I can work

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi