similar to: Need Advice/Suggestion

I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before?

Dear Uwe, (When replying to your message, I sent the reply to r-devel and not r-package-devel, as Martin Meachler suggested that this thread would be a better fit for r-devel.) Thanks. In the example below I used rm() explicitly, but in general users wouldn't do that. One of the reasons for the large number of file handles is that sometimes unnamed temporary objects are created. For

Hello, I have a problem with a comboot 16 bits program for syslinux, I use the API openFile, ReadFile and CloseFile for read my own config file. It's ok. I use the int13h to write a floppy disk (syslinux runs on a USB stick), after the first call to int13h, all the call to openFile return an error. Do you know what's the problem? Is it possible to use int13h when we use syslinux API?

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Dear Luke, Thanks. See below On 07-08-18 17:07, luke-tierney at uiowa.edu wrote: > In R 3.5 and later you should not need to gc() -- that should happen > automatically within the connections code. Could you elaborate on what has changed in R 3.5? As far as I can tell my problem also occurs in R 3.5 (my computer is still on 3.4.4; but I assume the solaris CRAN machine isn't). And

Hi All, We are using speex decoder only for narrow band decoding. Quite often, even though source frame is not silence, decoded audio is silence. I wonder there is a problem in the way we uses speex decoder API. Basically, we initialize speex decoder only once and just reset the decoder before we decoding another file. Do we need to intialize speex decoder and destory decoder for each audio file

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.

Hi, Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan "beep beep beep The person you are calling is unavaiable, please try again." I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no

In response to those who asked for a better explanation of what the Vilno software does, here's a simple example that gives some idea of what it does. LABRESULTS is a dataset with multiple rows per patient , with lab sodium measurements. It has columns: PATIENT_ID, VISIT_NUM, and SODIUM. DEMO is a dataset with one row per patient, with demographic data. It has columns: PATIENT_ID, GENDER.

Is there a way to read a file one line at a time? My file is potentially very large (multiple gigabytes) so I'd like to read and process one line at a time. The only choices I see are: 1) Direct the file to standard input and use readline(), but then I have to parse the data columns myself. 2) Call scan() with nlines=1 over and over again, incrementing skip, but I suspect that it will

Hello Everyone, I am receiving following error message will making Zaptel on Cent OS 5.2. make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel installed." You do not appear to have the sources for the 2.6.18-92.el5 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

hi all, In centos 4.4 turnoff monitor is not working.Dpms is enable and also change the dpms setting it working but not functioning. I am using kde3.3 desktop.How to slove this....pls help me. thanks, karthik -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Not sure if this is implied in the documentation or something given: does the RoR framework require "every table must have an id" ? If this is the case, then what happened to the concept of normalization ? Futher, if the tables have already been build for an existing appication ( without ID column in everytable ) , will the script / generate work ? Thanks -- Posted via

Hello Everyone, I have write a script following script for nagios -- typeset -i CRITICAL; #Positional parameter CRITICAL=`echo $2`; ME=`basename $0`; #echo $CRITICAL if [[ "$2" == "" ]] then echo NO INPUT!!! Usage ./$ME -c N else typeset -i ASCALLS; ASCALLS=`asterisk -rx "core show channels" | grep active | grep call | awk '{print $1}'` #echo $ASCALLS;

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, i want to replace a bunch of Apple Xserves with Xraids and HFS+ (brr) by Sun x4200 with SAS-Jbods and ZFS. The application will be the Helios UB+ fileserver suite. I installed the latest Solaris 10 on a x4200 with 8gig of ram and two Sun SAS controllers, attached two sas-jbods with 8 SATA-HDDs each und created a zfs pool as a raid 10 by doing something like the following: [i]zpool create

Hi folks, I've been playing around with an Asterisk server in my office for a few weeks now, and I've got it pretty much nailed down the way I want it, which is nice. One of the features I'm using is the ability to switch different contexts in & out of the dialplan on a schedule. So, for example, I've got the "official" tel number ringing my desk phone between

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the