similar to: PDC operation allowing guest machines to log in ?

Mm, I have a big Quad-Opteron.. thing.. that I''m trying to get Xen onto. I''ve used the 3.0.2 binary-install mode, updated menu.lst as per the README, but I need an initrd which contains the HP cciss RAID driver, and no Xen initrd image was installed into /boot. Now I notice xen-3.0.2-2-install/install/lib/modules/2.6.16-xen/kernel/drivers /block/cciss.ko But I

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like "You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes" I've seen on

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Hi :) I'm having enormous probs with the Budgetdedicated APT repo on Ubuntu dapper... 0.9.17 works a treat, but 0.9.18 and 0.9.19 immediately call the debugger on /any/ app, even winecfg + /usr/bin/notepad.. I've tried deleting my .wine dir, and always ensure there is no wineserver hanging around, but always to the same end effect: gdh@plip:~$ rm -r .wine gdh@plip:~$ notepad wine:

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case

It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons.... I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? Cheeres, Gavin.

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Today is my first day with sieve, so be gentle :) I'm trying to set up a pretty webmail interface to our Dovecot 1.2.4 server using roundcube. The managesieve config + roundcube 'managesieve' plugin work fine, and I'm able to use roundcube's UI to generate .dovecot.sieve files. We use winbind + LDAP lookups to do some exotic mail rewriting... ultimately user.name at domain.com

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Howdy, I would like to use the samba server as a replacment for NT Servers. So I need the add user script command for adding new machines to the domain. This is no problem in all 2.2.X versions of samba which I used. But on version 3.0 alpha and beta it is not working. I am using debian 3.0r1 with the unstable package of samba 3.0beta. smb.conf: [global] workgroup = unreal server string

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

Hi, I'm putting together a system to manage agents with Realtime, and without chan_agent. In 1.2.13, there's a handy (although marked as deprecated in apps/Makefile) PGSQL application to let me do this: macro queue-addremove(queuename,penalty) { switch(${MACRO_EXTEN:0:1}) { case I: // Login PGSQL(Connect connid host=XXX user=XXX password=XXX

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to

logins Reply-To: I've sourced the groups, but haven't had any success with solving this problem. Here are the details: I have a debian/testing system running samba 3.0.0beta2-1. It was working fine with the 2.x series, until I upgraded to 3.0. The debian server (al) is a PDC for a small NT domain (isabela) with three workstations, all running win2k pro. All but one work fine

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only