similar to: Centos really slow as VMware Server guest OS..

Hi, I am trying to set up a Fedora Directory server for centralised authentication. I configure the directory server, add a user called (via the Java GUI) test and then, using system-config-authentication, enable LDAP on both tabs. I then try to log-in using the test account I set up on the directory, but I get an error message in /var/log/messages: May 30 16:28:27 ds1 sshd(pam_unix)[4445]:

Does the nvidia drivers (downloaded from nvidia) support or work with the new centos 5 (beta) ? The version of X windows is different I think. Jerry

I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all.

Anyone had any experience with an Asterisk server as a VMWare virtual machine?

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: ***

Hi, I was wondering if we can check the voicemails remotely from a cell or a landline number. We have SIP 3 Digit Extensions connected to Asterisk server. If users are away from Desk & need to access voicemails can they dial in to Asterisk PBX & check their messages. I know one can check through web link & even have mailed. Aslo I have checked regarding DISA, but I am not kind of

I am soon going to be setting up a box to run VMware server but have to make the choice between Centos 4.x or 5.x.. I have tested both and VMware installs in both.. So are there any pros or cons to either version or does it make no difference at all? TIA

Hi, I guess the subject says it all.. I am thinking about combining a few servers onto a single server since they mostly do nothing much at all which would mean there would be less load on the UPS and a lot less noise in my office.. Since VMWare server is now free its probably the perfect time to do it.. The only issue is getting the VMWare up and running.. I would prefer to run it on

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.

I have a CentOS 4 server at home where we store all our data, I wanted to get some idea of the amount of traffic going into and out of the server by protocol so I installed IPtraf.. When I run it from an ssh session it looks fine but when I run it directly on the server console its doesn't look right at all.. It doesn't seem to layout correctly.. Is there something I need to do

Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that

Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and

Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector

Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. Thanks..

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

Hi, I am looking for a budget IP phone that can use preferably iLBC or GSM codecs.. Suggestions?