search for: besch

Hello, I would like to access my data frame without one variable. E.g.: > colnames(x) [1] "Besch" "Ang.m" "Arb.m" "i10" "Umsatz" "arbstd" I can try x[,-1], but this variable must be called by it??s name. x[,-"Besch"] x[,!"Besch"] attach(x) x[-Besch] ... ... does not work. I could not found a solution of this l...

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I coul...

...to go about this. 2) I don't have any means to distribute the files (other than e-mail, which I really don't want to to. So, if someone could brief me on the GPL issue, and (perhaps someone else) offer a distribution point, it's free for the asking, VB sources and all. Stephen R. Besch

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

...curiity, but since the phones are all on 192.168, I don't think it is too much of a risk. However, if I was on a public network, or, if the phones were using DHCP for addressing, this could be a major headache. Has anyone else had this problem and if so, what was the solution? Stephen R. Besch

Mich besch?ftig folgende Fragestellung. Ich kenne die Verteilung (lognormal) zus?tzlich weiss ich das 99%, das 90% und das 1% Quantil. Gibt es in R eine M?glichkeit die Lognormalverteilung zu finden, das heisst den korrespondierenden logmean und logsd? Vielen Dank f?r ihre Hilfe Gruss Yvonne

The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through

...hich is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as it's already appended by the script. Hope it's useful.. Thanks to Stephen R. Besch for information about the format of this file ! (One thing I am not 100% sure of: do I have to append zeros to the end of the body until it has an even amount of bytes, or an even amount of words ? Right now, I do both.) Regards, Leon de Rooij -------------- #!/usr/bin/perl -w use strict; my...

Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the zaptel and asterisk compiles seg fault. I am assuming that they will compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 compiled and is now running. Is there

...uixun wu) 7. bristuff 0.0.3 ? (Bjoern Adler) 8. RE: VoicePulse changes (daryl@introspect.net) 9. RE: freenode #asterisk IRC channel identd problem (Mike Reed) 10. Re: freenode #asterisk IRC channel identd problem (Steven Critchfield) 11. Re: Updated Grandstream configurator (Stephen R. Besch) 12. Re: Updated Grandstream configurator (Stephen R. Besch) --__--__-- Message: 1 Date: Thu, 15 Jul 2004 12:39:45 -0700 From: echo-asterisk@secondphone.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Bounty! For help with echo cancellation code. Reply-To: asterisk-users@...

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

...s, I wonder if anyone has found a more elegant solution, remembering that I want to give some extensions access to only the local context, but still provide toll free service for everyone (i.e, I don't want to move the "_918" definitions into the longdistance context). Stephen R. Besch

Hallo Liste, hat sich schon mal jemand mit 2 Faktor Auth f?r dovecot besch?ftigt? Es geht dabei nicht nur um die Absicherung einer Weboberfl?che sondern direkt die Absicherung der Standard pop/imap Verbindungen des Clients. Wir machen das auth bisher ?ber MySQL.? Rein technisch w?rde das gehen, da man die Password-Query beliebig anpassen und auch mit allow_nets arbeit...

...there isn't. The "blinking" phone >designers should have put those stupid blinking red leds - that only >flash on boot up - under the message button and flashed the display >during boot up. But they didn't and we're stuck with it. Such is life. > >Stephen R. Besch I noticed this strange factoid as soon as I got MWI to work. Does anybody know what then is the use for the message button? Thanks, Aldo

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

From what I understand, shared interrupts are permitted on the PCI bus. Whether they work for a specific card or not depends entirely on how the driver is written. Someone at Digium should know this. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106