similar to: [Fwd: Re: Re: OpenBSD/i386 on Xen 3 with HVM]

Thanks --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0650-2, 23/11/2006 Tested on: 26/11/2006 1:27:17 PM avast! - copyright (c) 2000-2006 ALWIL Software. http://www.avast.com _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Hi. Can you give me some hints on how I can create a rotational board? I dont even know how to spell it in english. What I want is to have more than one line reserved, but with a single phone number, so that people can call to the same number and get a ringing signal if any of the lines is available, instead of having to dial 5 different numbers in order to get a free line. This is done

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

[Bonsoir] Sorry for the first post with a bad adress ... :o( good evening [ I'm french and my english is so bad ... so ...] I use pxelinux with vesamenu.c32 and it's really good but ... I would like to use correctly some carracters with accents I try severals psf files found here and there but these fonts dont have graphicals carracters used for borders of the menu. So... two

Hi, I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. Thanks. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0436-4, 03/09/2004 Tested on: 07/09/2004 13:57:36 avast! - copyright (c) 2000-2004 ALWIL Software.

Hi again. My last post was incomplete, so I' reposting it. I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. I'm interested in buying a TDM card (probably a TDM11B card) and I need to know if it will fully work on Spanish PSTN and what

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

Evening folks: I did an install.views of finance and econometrics and between the two of them R reported 'FracSim', 'RDCOMClient', 'snow', 'VGAM', and 'segmented' as missing dependencies. Now, I've managed to hunt down what appear to be current zip file copies of all of the packages but 'snow'. Is there a windows edition of the package out

Hello, I've tried NUT trunk SVN branch some time ago in order to use it with a Blazer USB UPS (megatec protocol driver supported by Carlos Rodriguez). This UPS wasn't yet supported by NUT but I was able to compile and test the software on my Debian. I download in order to try latest Rev: 805 today to see if USB support for my UPS where better but the 'autoreconf' command give me

Afternoon folks: I'm getting a program crash when I try to run demo(rgl). The following error details result: RGUI caused a stack fault in module NVOPENGL.DLL at 017f:695280f0. Registers: EAX=00000002 CS=017f EIP=695280f0 EFLGS=00010246 EBX=00000001 SS=0187 ESP=00572000 EBP=004d1208 ECX=042f1f01 DS=0187 ESI=004d1208 FS=5d1f EDX=00442d84 ES=0187 EDI=042f1f3c GS=5d0e Bytes at CS:EIP: 53 56 8b

Hi all, Until last night my XP Pro clients and my Samba PDC (2.2.3a) on RH7.3 existed quiet happily together once I'd changed the requiresignandseal to off and made the other changes. Last night I made a change to the config which failed so I rolled back to my previous working config file. Now my XP laptops can not connect to the domain - they report can not find domain controller or can not

Peter, I have been doing some more testing and I find that if I modify the libusb.c file to set a report size in libusb_get_report to 16 the error messages go away. This is the code as it currently stands: static int libusb_get_report(usb_dev_handle *udev, int ReportId, unsigned char *raw_buf, int ReportSize ) { int ReportSize_test = 16; unsigned char raw_buf_test[ReportSize_test +

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi. I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any of the 4 PSTN lines connected to the asterisk box from and dial your number, and asterisk will dial out through one of the 4 sip accounts I have on a SIP -> PSTN provider. I think of something like this in the extensions.conf [incoming] exten => s,1,Wait,1 ; Wait a second, just for

[Note: trimmed HTML] ---------- Forwarded message ---------- From: selinger@mathstat.dal.ca (Peter Selinger) To: jon.gough@eclipsesystems.com.au (Jon Gough) Date: Sun, 28 Jan 2007 02:20:48 -0400 (AST) Subject: Re: [Nut-upsuser] Ablerex 625L USB version Hi Jon, I can't figure out which version of NUT you are running, because you have pruned that information from your output. Also, which

> Message: 7 > From: "Glover George" <dime@gulfsales.com> > Also, as a side note, maybe someone might have the answer to this as > well. Once I switched to a domain, the user can no longer > open Windows > Messenger in XP. Is this normal? IS there some setting I need to > change? (i.e., is Messenger looking for an exchange server > only when in >