search for: alterars

Cordial saludo. Estoy teniendo un problema al convertir una columna categórica a numérica. Espero que me puedan colaborar. Esta variable está en números escritos, es decir, "uno", "dos" etc. Ya los cambie por números (que siguen siendo factores) mediante fct_recode y quise de ahí convertirla mediante as.numeric pensando que R iba a identificar los números y no iba a alterar el

Hi people, What are the pros and cons of Winbind, LDAP and Kerberos in a Samba 3.0.2 plus Active Directory environment ? What could be the best design for this scenario ? Estevam Henrique ========================================================= Esta mensagem pode conter informacao confidencial e/ou privilegiada. Se voce nao for o destinatario ou a pessoa autorizada a receber

Hi all, I recently updated my to R 2.6.0 and tseries package ?tseries? version: 0.10-11. When i was using R Version 2.3.1 (2006-06-01) with tseries 'tseries' version: 0.10-7, the code > garch(dflnRCLC1) ***** ESTIMATION WITH ANALYTICAL GRADIENT ***** Call: garch(x = dflnRCLC1) Coefficient(s): a0 a1 b1 4.985e+00 1.880e-01 6.210e-14 > worked very

You should use the ntlm_auth module provided by samba. -----Mensagem original----- De: samba-bounces+ecarvalho=bmf.com.br@lists.samba.org [mailto:samba-bounces+ecarvalho=bmf.com.br@lists.samba.org] Em nome de Riccardo Baldanzi Enviada em: quinta-feira, 22 de abril de 2004 16:51 Para: samba@lists.samba.org Assunto: [Samba] Problems with ntlm_auth --helper-protocol=squid-2.5-ntlmssp Hi Guys,

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi. I have some apps that i have "admin" areas to the clients manager the app. This "admin" is a directory on my controllers path, like this: my_site --> app -->-->controllers -->-->-->admin And my controllers in "admin", i use the correct namespace(example for clients_controller): class Admin::ClientsController < AdminController Only doing

Hi Jim, I did what the doc says but the problem is the same. Does anybody saw this work ? I mean, is the Samba 3.0.2a+Kerberos MIT 1.3.3 able to be accessed by a WXP, W2K or W2K3 machine, using Kerberos tickets generated in a Windows 2003 KDC (W2K3 AD) ? Thanks -----Mensagem original----- De: Jim McDonough [mailto:jmcd@us.ibm.com] Enviada em: segunda-feira, 19 de abril de 2004 17:07 Para:

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hola, Llevaba tiempo pensando en lanzar esta pregunta a la lista, sobre "¿Cuál es vuestro uso de R?". Mejor que colapsar la lista, además de que seguramente por el tipo de respuestas (no estructuradas) complicaría su análisis, he decidido crear una encuesta online aqui: http://www.tuvotacion.com/cual-es-tu-uso-de-r La encuesta la he dividido en estas tres opciones básicas, que confío

I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql?

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

Hi, I'm wondering what I need to change to get the "swedish" type ring on a SIP-trunk. When I make an inbound call i still have the "US"-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan

Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop?

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

hello my friends, when i make a genzaptelconf i get this message ******************** CAS signalling on span 2 conflicts with HDLC with FCS check on channel ******************* Any idea Please? I m installing zaptel 1.4 i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb