similar to: Meetme and authentication

Hello, I have one UPS Infosec XP Pro 2500 RM and it works with windows sw RUPS2000. Now I want to install this UPS as virtual server on vmware ESXi vMA - (SUSE SLES 11SP2 linux). On this linux I have NUT 2.6.0 and I stopped now with configuration of genericups driver. Working cable (direct cable pin1-pin1 ... pin9-pin9) connection on Windows: PIN | PC | UPS

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

My thoughts on this after reading Steven's very politically worded reply is that IMO your best bet would be to go with RedHat, I am not going to go into details about the if's, when's, why's, and but's.. I am running Asterisk quite well on RedHat 9 and if you like I have created an install guide for setting up an Asterisk box on RedHat 9 which I can send to you if you are

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

[please use reply-all when responding - the list does not modify the reply-to header.] > On Nov 16, 2015, at 8:24 AM, R?bert SZAKSZ <robert at szaksz.eu> wrote: > > Hello, > > I have one UPS Infosec XP Pro 2500 RM and it works with windows sw RUPS2000. > Now I want to install this UPS as virtual server on vmware ESXi vMA - (SUSE SLES 11SP2 linux). > On this linux I

Hi Charles, >----- Original Message ----- >From: "Charles Lepple" <clepple at gmail.com> >To: "R?bert SZAKSZ" <robert at szaksz.eu> >Cc: nut-upsuser at lists.alioth.debian.org >Sent: Monday, November 16, 2015 3:20:17 PM >Subject: Re: [Nut-upsuser] Contact-closure UPS > >[please use reply-all when responding - the list does not modify the

>On Nov 20, 2015, at 12:17 PM, R?bert SZAKSZ <robert at szaksz.eu> wrote: >> >> No progress has been made: >> >> cable is UPS - PC >> 6 - 3 >> 7 - 5 >> 9 - 2 >> > >Maybe pins 4 and 7 (ground) both need to be connected on the UPS side? I connected both to pin 5 >Unfortunately, I don't think there is a safe way (in software)

On 18 April 2016 at 16:33, Silviu Baranga <Silviu.Baranga at arm.com> wrote: > Doing a grep "eabi" * -R | grep darwin in llvm I found the test divmod-eabi.ll > which uses the triple armv7-apple-darwin-eabi. What format does that have? Certainly not ELF. :) But I didn't mean "has eabi on triple", but "is in none-eabi mode", which may have to check a

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

On 18 April 2016 at 16:18, Silviu Baranga <Silviu.Baranga at arm.com> wrote: > This doesn't look like something ACLE specific (I can't find it in the ACLE doc). Sorry, I didn't mean it was ACLE, only that you guys were fiddling with macros. :) > This seems to be a generic macro. I think it would make sense to define it > if we know we're emitting ELF. Since the

I''m playing around with foreman for the moment. I can''t seem to figure out how to make the dashboard look like in this screenshot http://theforeman.org/wiki/foreman/Screenshots#Dashboard In order for those statistics to work what should be done? I have activated the rrdgraph reports in puppetd, uncommented the line ":rrd_report_url: report/" in config/settings.yaml

> On Apr 29, 2015, at 2:33 PM, Matt Arsenault <arsenm2 at gmail.com> wrote: > >> On Apr 29, 2015, at 10:06 AM, Silviu Baranga <Silviu.Baranga at arm.com <mailto:Silviu.Baranga at arm.com>> wrote: >> >> Note that dividing by an integer constant should be a cheap operation >> compared to FP multiplication and comparison as this would get lowered to a