similar to: Server for 100 concurrent calls

hello, everyone there is anybody can help me, what is the address of the samba lastest release samba 3.0 samba 2.2.x because i wanna upgrade my current samba server 2.2.3a

Hello. I am administrating about 30PCs with W2K and users are authenticated in SAMBA running as PDC on Linux. About 16PCs have the possibillity to boot also into Linux. I would like to authenticate also linux users using the samba, because I do not want to copy passwd, group, shadow each time I add a user, or even add the user on each PC. I do not use LDAP a do not know NIS. Is it possible to use

as.character.factor contains this line (where cx=levels(x)[x]): if ("NA" %in% levels(x)) cx[is.na(x)] <- "<NA>" Is it possible that this is no longer the desired behavior? These two results don't seem very consistent: > as.character(as.factor(c("AB", "CD", NA))) [1] "AB" "CD" NA > is.na(.Last.value)[3] [1] TRUE

My wife has just gotten a new printer which works perfectly when connected either to her Win98 box or to my woody box. However I am unable to share the printer between the two machines. Here are my choices in order of increasing desireability. I could of course just move the cable whenever I want to print from my box. This is the obviously least desireable option. The windows machine can

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75

i had a sip invitation problem with my voip provider and here the message that was shown : Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:XXXXX@195.112.214.99>;tag=as7b43dfbd' -- SIP/callshop-3fcc is circuit-busy == Everyone is busy/congested at this time -- Got SIP

Dear R developers, i have already send the question below to r-help but got no responses. Perhaps it is more suitable for r-devel due to its rather technical level. It would really help me to find a solution (or to find out that there is none). Is there any way to access/print/save the content of an environment in which an error occoured? Imagine the following testcase: test =

Hi, I have just started using Dovecot (1.0.10) SASL behind my Postfix (2.3.3). Everything works fine, as long as I send the SMTP AUTH command manually. When using M$ Outlook (2000 and 2003) and Outlook Express 6, authentification always fails. Turning on the Dovecot auth debug on reveals the reason: Dovecot SASL expects, in accordance with the standards, the following format of the SMTP AUTH

Hi all, I've recently upgraded to red hat 7.2 and found that samba no longer starts up with the rest of the system. Thanks to another member on this list I know the command to start it manually is /etc/rc.d/init.d/smb start but where do I have to put this command to get it to run at startup with the rest of the services? Thanks

I rebuilt an RHL 7.2 and copied my Samba file(s) back over. Samba is running, and when prompted for my username/password by my Windoze machine it will only work with one username/password. It will not work with any others. What's causing my problem here? Don't shoot the messenger, Rick Reed rreed@budgetext.com

Hi, I've dug a bit through the wiki and the mailing lists, and haven't really seen anything like this, but there must be someone out there doing this. Basically, there is a fax line that I don't want to answer inbound, but I want it available to do dial out from. Right now, we are using a busy wait around the ringing line, but I was hoping for something that might be a little more

Hi, Can someone explain to me how ATAs operate in a call shop type environment to provide realtime billing to the callshop software. The ATAs seem to be configured to connect directly to the asterisk server, so how does the call shop software report in realtime your destination and duration of the answered call? Thank you, AC -------------- next part -------------- An HTML attachment was

This is pesky; cant get on net yet with brandy new ubuntu, as cannot install modem and no clear directions in ubuntu 10.04 need wine: cannot find how to get: all net pages assume youve got modem up and running ie advising just use "downloader" or "package source"....what about no modem yet in linux? (looking here thru windows)... impossible Need to download wine for Ubuntu

Hi all, I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my Asterisk@Home. Just want to clarify some licensing issues regarding them. If i had to do a full reformat of my PC and reload a@h again will i be able to use the licenses again without re-registration? If no. .Is there are limits for this? Please anyone clarify. Thanks

Thers a box we can get here in Australia that has an RJ45 plug, a built in web server that has a config page and URL's to close/open one of 4 included relays. I use phpagi to hit the url and open/close doors that way. If anyone is interested, ill let you know URL's but its being sold from our local Jaycar agent. Regards Kevin -----Original Message----- From:

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- -------------------------------------------- Paulo Garcia Pika

Hi! I am new to asterisk and I'd like to know wheter the following scenario is possible: Someone press the Button on the door station. The door station dials lets say the extension 333. I take the call on 333 and talk with the person on the door. Now I'd like to activate the door opener by pressing some numbers on the analog telefon. Asterisk should now recognize that I pressed