search for: rememeber

...often after I closed all R windows, my CPU usage was still 100%. By checking the task manager, I found there are one or several "Rgui.exe" still running and took all the CPU. I had to close them one by one manually. This happened to me with R 1.6.1, R 1.6.2 also and also on Win2K. Rememeber there was a "memory leakage" problem with the early release of 1.6.2. Is this what I'm exprencing here? Or this is due to I'm runing R under SDI mode, b/c my colleague hasn't found the problem with his R 1.7.0 or other versions which are runing under MDI mode. Help........

I rememeber seeing an article about how to cope with legacy databases where the primary key is defined as VARCHAR2. I only want read-access to the data. Does anyone have a link to such an article? Or can give advice? -- Posted via http://www.ruby-forum.com/.

Hi, I have 2 internet connections (1 adsl/1 cable). I am try to route all outgoing mail from the mail server (on the same box), through the ADSL connection routing through the cable will mean mail will get rejected by AOL :( I am using qmail as the mail server. The configuration is: eth0 : cable connection ppp0 : adsl connection eth2 : internal lan connection I have configured split access as

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

...Of Kanuri, Seshu (Company IT) Sent: Monday, June 06, 2005 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to make Polycom phones work with Asterisk as aSIP Client? Folks! If you are using Polycom Phones, any model - 500, 300, 400, 600 etc, please rememeber to add this line to your sip.conf entry. Progressinband=no Without this line, these phones may not work. Probably this one line may fix most of the problems users are reporting on this forum about Polycoms. Seshu Kanuri -------------------------------------------------------- NOTICE: If receiv...

I am the network administrator at a small (20-30 employee) financial company. We are in the process of moving offices and will be obtaining a VoIP phone system when we do. Right now, it's down to the 3com nbx100 series and *. Having lurked on *-user for a few weeks and having seen the nifty features of asterisk, I'm convinced. The price difference has pretty much sold my superiors.

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

...ression=yes ; ; List all devices we can use. Contexts may also be specified ; ;context=local ; ; You can set txgain and rxgain for each device in the same way as context. ; If you want to change default gain value (1.0 =~ 100%) for device, simple ; add txgain or rxgain line before device line. But rememeber, if you change ; volume all cards listed below will be affected by these values. You can ; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%). ; ;txgain=100% ;rxgain=1.0 ;device => /dev/phone0 ===================================== =========================================...

I have a file named "test.csv" with the following 3 lines: %y-%m-%d;VALUE 1999-01-01;100 2000-12-31;999 > read.table("test.csv", header = TRUE, sep = ";") delivers: X.y..m..d VALUE 1 1999-01-01 100 2 2000-12-31 999 I would like to see the following ... %y-%m-%d VALUE 1 1999-01-01 100 2 2000-12-31 999 Note, >

I would like to start a discussion centred around the various ways one might serve up configuration files from an Asterisk server (I know, it's better to use a secondary server for all this, but let's talk about a smaller system). The types of things being served would include: - Logo image for sets that support that - XML directory files - XML or raw text configuration files -

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

Hi, I''ve added some preliminary support for VT-d for paravirtualized guests. This must be enabled using an ''iommu_pv'' boot parameter (disabled by default). I''ve added some python bindigs to allow xend to assign PCI devices to IOMMU for PV guests. For HVM guests this is handled in ioemu. Not sure if it makes sense to handle both cases in one place. The

I run a small Linux webserver and NAT router from my cable modem at home. Whenever someone starts an http download, all other traffic from my LAN is starved. Bandwidth is not really an issue, but latency is particularly horrible -- pings that usually come back in 20ms can take up to 600ms while the web server is active! I set up QoS (netfilter+iproute2) on the NAT machine in an attempt to give

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.