search for: wieling

...esult in an actual decrease of the degrees of freedom invested by the total model (caused by a decrease of the edf's of splines in the model with the interaction). In this case, how would I proceed in determining if the model including the interaction term is better? With kind regards, Martijn Wieling -- ******************************************* Martijn Wieling http://www.martijnwieling.nl wieling at gmail.com +31(0)614108622 ******************************************* University of Groningen http://www.rug.nl/staff/m.b.wieling

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

...ard for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date: Mon, 25 Jul 2005 08:31:47 -0500 From: Eric Wieling aka ManxPower <eric@fnords.org> Subject: Re: [Asterisk-Users] Need Advice To: tarsi@binhost.com, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <42E4E9C3.2080806@fnords.org> Content-Type: text/plain; charset=us-ascii; format=f...

Hi everybody, I got a problem here with my Samba ADS member server. It functions as a file server and works for a long time. A few weeks ago our Active Directory domain controller failed due to power outage. We fixed it and now an employee reports that he could not use the samba file server any more. I restarted samba service and did the following testing: 1) getent passwd <thisUser>

On Tue, 26 Dec 2017, Eric Wieling wrote: > Don't use an 'h' extension, use a hangup handler.?? Why? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/...

I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never

...iguration: single image plane Page Number: 0-0 Group 3 Options: EOL padding (4 = 0x4) Notice the Page Number: info. I've heard there's a lot of software out there that can't view multi-page TIFFs. I don't know if that's what I'm experiencing or no. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

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

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

...4 bit version in Unix) and mgcv (both version 1.7-10 and 1.7-6). In my (simplified) model predicting pronunciation distance I'd like to include a random slope per Participant for a binary value (IsDem) which stores a word-specific characteristic. I load the data (available at http://www.martijnwieling.nl/dat.csv) and run the model as follows: > library(mgcv) # version 1.7-10, but problem also occurs with earlier versions (e.g., 1.7-6) > dat = read.csv('dat.csv',header=T) # data available at: http://www.martijnwieling.nl/dat.csv > dim(dat) # the original dataset is larger, but t...

...r by-word random slopes than words, and/or ii) how I can link the random slopes which are present to the correct labels of the words? (I did not include the model as it is >300 MB in size, but let me know if this is necessary.) Any help would be greatly appreciated! With kind regards, Martijn Wieling University of Groningen http://www.martijnwieling.nl

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

OK before i start, this was something i was just testing while everyone was in training this afternoon. I have never had to deal with subnets before as we only have a few users, but decided to do a little bit of messing around to gain some experience. Basically we have one server (mandriva 2006) running samba, it has two 4 port ethernet cards in it but only utilitising one port from each card.

Hello! I'd like to Flash() a modem line (BRI) with Asterisk. It is a passive ISDN-card connected to a hardware PBX. I use ISDN4Linux. I recognised that unfortunately the Flash() application flashes Zap devices only. Now I am wondering how I could flash Modem/ttyI0. The source code chan_modem.c doesn't contain anything about flashing a modem line. So I tried to simply put the AT-command

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: Thursday, December 14, 2006 4:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Hardware TDM Switching > > asterisk@nicox.org wrote: > > > Do anybody know, if there is a way to connect 2 zap-channels with > >...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi, I'm using Samba 3 on a Debian 5.0 (Lenny), the package is version 3.4.7 (from Debian backports, the full version 2:3.4.7~dfsg-2~bpo50+2). We use some 'veto files' on our shares to avoid users to have multimedia files (movies, mp3 and so on). Now we have a true demand to allow certain users to store corporate videos (about our

There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm