similar to: NT4 roaming profiles not being saved completely

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:

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

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 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

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

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 :) 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

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

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

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:

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

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

Ok, did some experiments. Here's the scoop. You will need a live USB key of CentOS 6 with a persistence layer (overlay) and the EPEL gdisk package installed to make this thing boot. On a system with Chameleon already installed, boot the CentOS 6 install media. Installing Chameleon without OSX is an adventure that I've not done, so a 'testing' OSX install (10.6) with the

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.

Hello, I have a little trouble when entering into a domU console (xm console mydomU) : i can''t use the CTRL-C sequence to stop a script/command (like a continuous ping for example) Is there any parameter / tip for that ? Arnaud _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

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});

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

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our