search for: abernathy

On 8/17/2020 6:55 AM, Rowland penny via samba wrote: > wbinfo --group-info 'Domain Users' | awk -F ':' '{print $3}' > > wbinfo -i <username> | awk -F ':' '{print $3}' > > Rowland > root at mbr04:~# wbinfo --group-info 'Domain Users' | awk -F ':' '{print $3}' 10000 root at mbr04:~# wbinfo -i [username] | awk -F

...ot at mbr04:~# wbinfo -i [username] | awk -F ':' '{print $3}' > failed to call wbcGetpwnam: WBC_ERR_DOMAIN_NOT_FOUND > > Could not get info for user [username] Hmm, I only get that if I ask for the info for a user that doesn't exist: rowland at devstation:~$ wbinfo -i abernathy | awk -F ':' '{print $3}' failed to call wbcGetpwnam: WBC_ERR_DOMAIN_NOT_FOUND Could not get info for user abernathy Are you sure that 'username' exists in AD ? If you run 'wbinfo -u | grep username' , do you get output ? Rowland

I am new to the mailing list, but I am very interested in running my small home business office phone system using Asterisk. However, Broadvoice, a VoIP provider of choice based on my research, is not available in my area. I currently use Vonage VoIP. Their website mentions nothing about being able to link to Asterisk. I was wondering if any US subscribers have been able to configure Vonage

...rint a banner page for each job sent to it through normal solaris machines. I disabled this and still the two header and trailer pages continued so I'ver pretty much exhausted my knowledge on this matter. Does anyone know how to get rid of the header and trailer pages that samba creates. --B.A.Abernathy

I'm trying to use a Nas4Free storage system. This same hardware worked fine on Freenas 0.7.2. O/S: Nas4Free 9.0.0.1.148 which comes with nut 2.6.4. UPS is an APC Back UPS RS 800. USB interface. Driver : usbhid-ups Port: auto my NAS is communicating with the UPS, but in testing power failures, I'm not getting the shutdown to be handled properly. When I pull the AC plug on the UPS,

...sible fraud? I might be able to help with such an effort. I have NOT done much with election data, but I have developed tools for data analysis, including web scraping, and included them in R packages available on the Comprehensive R Archive Network (CRAN) and GitHub.[1] Penny Abernathy, who holds the Knight Chair in Journalism and Digital Media Economics at UNC-Chapel Hill, told me that the electoral fraud that disqualified the official winner from NC-09 to the US House in 2018 was detected by a college prof, who accessed the data two weeks after the election.[2] Spe...

I don't see anything on the Wiki or in the documentation about disabling this feature.

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

Yes, After call transfer,I don't want to be media go through Asterisk. Is it possible ? Thanks, Karun. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Dameon D. Welch-Abernathy Sent: Thursday, August 12, 2004 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Blind Call Transfer using Sipura 3000 + asterisk On Wed, 2004-08-11 at 12:00, Karun Chemudugunta wrote: > I am using Sipura 3000 to receive PSTN calls and forward those calls to >...

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

...ongress to use its authority now to shape these important communications policies impacting constitutionally based copyright laws and First Amendment liberties. Signed, (Affiliations for individual signers are for identification only. Endorsing organizations are listed separately.) William Abernathy, Independent Technical Editor Scottie D. Arnett, President, Info-Ed, Inc. Jonathan Askin, Pulver.com John Bachir, Ibiblio.org Tom Barger, DMusic.com Fred Benenson, FreeCulture.org Daniel Berninger, VON Coalition Eric Blossom, GNU Radio Joshua Breitbart, Media Tank...

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

I'd like to announce the time when people call and hit my voice-menu context, but the complaint is that the time announcement isn't interruptable. Is there any way to make SayUnixTime interruptable? -- PhoneBoy