search for: steveh

Dear all, I am trying to find a R version of the S+ routine fac.design. In the help of the package conf.design fac.design is listed under "see also". Any clues as to where (which package) I can find fac.design? thanks Steve. Steven Higgins Department of Ecological Modelling UFZ - Centre for Environmental Research Leipzig-Halle Permoserstrasse 15 D-04318 Leipzig Germany Phone:

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Hi, I'm writing a utility that needs to smbmount various shares from servers in numerous domains (no problem, all working) and then list the contents of the directories (no problem again) and obtain the windows file owner in a textual form..... Any ideas how I can achieve the last part efficiently? I see that smbcacls can do it 1 file at a time, I really need a way of doing it

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hello All, I am new to Dovecot and the Dovecot mailing list. I am trying to migrate an existing Exim + qpopper - mbox (pop3 only) installation over to Exim + Dovecot using mdir (pop3 and imap) And am running into problems with preserving the XUIDL headers. I am running Dovecot 2.0.14 I have used the mb2md.pl script : http://dovecot.org/tools/mb2md.pl from this page

That sound's like the right thing to do, you'd probably have contexts related to what phones could do "unrestricted","localdialling","extensionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension

I'm going to modify the queue announcements to allow for rounded seconds (e.g. we want to know to the tens of seconds. E.g. Average wait 1 minute 20 seconds). I'm going to add the optional announce of seconds to the queue config and a rounding factor (e.g. 10 in our case). The following parameters will be added Queue-announce-seconds (default is off) Queue-seconds (default

It'll work, either as a SIP phone with the SIP image, or as skinny using wither chan_sccp or chan_skinny (check the wiki). Steve -----Original Message----- From: Matt Davies | MattDavies.Net [mailto:matt@mattdavies.net] Sent: 02 July 2004 15:46 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960G and * I have been doing so much reading on phones lately that I have

It's so you can have agents that are less likely to take calls (e.g. imagine a sales queue, you'd have the sales people with no penalty, you might have the receptionists with a penalty of 1 and us propeller heads in technical support with a penalty of 2). The technical support people would only be offered a call from the sales queue if all the sales people and the receptionists were busy.

Hi, We're using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then we'd use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel:

Are there any other GDK users out there with Asterisk? I've got all the integration working, except voicemail. Does anybody know a way of disabling the forward to voicemail on a per extension or per DDI basis (I can disable the voicemail hunt group but then I can't light the MWI indicators as it seems that only ports marked in the voicemail group can issue the MWI on/off commands).

Can anybody see a way of detecting the current number of retries remaining to a call file in the extension context that it is calling? E.g. If I want to schedule a fax and I want to feed an email back to the sender stating that the number is busy 2/5 retries remaining? Steve The information contained in this email is intended for the personal and confidential use of the

Hi, Can someone explain how to use CVS to keep in sync with samba updates? I tried following the CVS_ACCESS.txt file, but it always seems to be behind the "current" release of samba. For example, right now if I do cvs -d :pserver:cvs@samba.org:/cvsroot login cvs -d :pserver:cvs@samba.org:/cvsroot co samba I end up with 2.0.0beta1. But 2.0.0beta2 has already been released.

Dear Samba list, I'm having a problem with browsing samba 2.0.0 from an NT Workstation (4.00.1381). When you try to browse the samba server, it returns with the error "\\machine\is not accessible. The remote procedure call failed." On the other hand, browsing works from a win95 and win98 machines. Printing and mounting shares work fine though if you explicitly specify the share.

Hi, Is it possible to make only the IPC$ guest ok so that you don't have to have to specify a password? In other words, I'd like 'map to guest'=bad user for all shares *except* IPC$, but to allow any user to browse the share list (ie: map to guest=bad password" for IPC$ only) Thanks.

Throwing errors relating to utils.h: /usr/include/asterisk/strings.h:264: parse error before `__extension__' /usr/include/asterisk/strings.h:264: parse error before `;' /usr/include/asterisk/strings.h:264: warning: type defaults to `int' in declaration of `__retval' /usr/include/asterisk/strings.h:264: `__len' undeclared here (not in a function)

[vested interest: I reported that bug!] I agree with this, a system that has x bugs still has x bugs even if you close them, I think they should be closed by houskeeping if they can't be reproduced and the person that reported them can't provide a way to replicate, but they shouldn't be closed because nobody has fixed them for a while. This seems to be the case on a number of

Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain

We're using Cisco phones running skinny protocol. When I call other extensions I don't get a ringtone, although the remote end does ring and when answered we get clear two way audio. When I call a queue from a skinny phone then I don't hear the announcements. Likewise we don't hear music on hold on these phones, although we can see mpg123 in the process list and ls -l the fd