similar to: drbd update 8.3.8.1

Hi All, I have what I believe to be a pretty basic LVM & RAID setup on my CentOS 5 machine: Raid Partitions: /dev/sda1,sdb1 /dev/sda2,sdb2 /dev/sda3,sdb3 During the install I created a RAID 1 volume md0 out of sda1,sdb1 for the boot partition and then added sda2,sdb2 to a separate RAID 1 volume as well (md1). I then setup md1 as a LVM physical volume for volume group 'system'. I

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

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 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

[ Please keep me in Cc: as I am not subscribed to this list ] Hello, we have noticed the following bug in rsync (including latest CVS snapshot): if I am copying from the remote host file that is unreadable by my account, the error is reported properly, but the errorlevel returned is 0. $ rsync distanza:/etc/shadow . send_files failed to open /etc/shadow: Permission denied rsync error:

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

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

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

Hi all. I have two hosts with drbd: kmod-drbd83-8.3.8-1.el5.centos drbd83-8.3.8-1.el5.centos and kernel (CentOS 5.7): 2.6.18-308.4.1.el5 After a recent upgrade of kernel I have had two sitiuations when my ext3 filesystem on /dev/drbd0 became read-only. I've checked disks with smartctl -t long, they are ok. There are no messages with disks problems in /var/log/messages | dmesg. I've made

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

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

Hi Everyone, I'm new to programming R and have accomplished my goal, but feel that there is probably a more efficient way of coding this. I'd appreciate any guidance that a more advanced programmer can provide. My goal -- I would like to find the length of the longest word in a string containing many words separated by spaces. How I did it -- I was able to find the length of the