search for: rememeb

Displaying 14 results from an estimated 14 matches for "rememeb".

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2003 May 03
2
Memory leakage?
...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......
2006 Apr 13
2
Legacy database with varchar2 primary key?
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/.
2004 Jan 22
4
Problems with netfilter
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
2003 Sep 04
4
update re. Grandstream + SIP + Echo problems ..
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
2005 Jun 06
0
How to make Polycom phones work with Asterisk asaSIP Client?
...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 rece...
2004 Jan 07
2
Asterisk success stories in small-medium office environments?
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.
2003 Aug 04
4
SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers
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
2006 May 01
1
unable to set outgoing callerid
...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 ===================================== =======================================...
2006 Feb 14
9
read.table
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, >
2005 Mar 05
6
Survey: what's the best HTTPd/TFTPd/FTPd to serve up configuration files to sets
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 -
2004 Oct 06
10
Eezee phone?
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
2008 May 19
21
[PATCH 0/5] VT-d support for PV guests
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
2000 Nov 18
9
priority bands don't reduce interactive latency?
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
2003 Aug 20
13
VoIP dialtone?
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.