similar to: 2 servers via PRI

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! 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

[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

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

Hi All, Greg has been a huge help getting me going with this tellabs echo can, but I'm still having some problems getting it to work... I suspect I wired it up incorrectly, so I thought I'd see if anyone can point me in the right direction. Digium tech support pointed me to this doc for a standard T1 cable: http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm which looks like it

Hello Dan, Have a look at this link: http://www.adcomcorp.com/asterisk/tellabs I got those pictures up there, may be of help. In essence, 1 pair is either a tx pair or an rx pair. If I recall, Orange and Green should be one side, and blue brown should be the other side. I tried to upload them to the wiki (even 40% size) but they didn't show up for some reason... Files are large on the

Hi all, I have thoroughly read the available documentation and I can't seem to find a workaround for my setup... I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain users). I use Asterisk 1.2.6 The available conferences are defined as follows: conf => 1000,user pin1, moderator pin1 conf =>

To All If anyone can shed any light on this it would be greatly appreciated. My phones are unable to enter pins numbers correctly when required by the party they are calling. For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid="Telx 7501" <7501> mailbox=7501@telx.com host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially "dtmfmode=rfc2833" What should my "DTMF Payload Type:" be set to on my SNOM 190 phone.

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

I am looking for a good Asterisk GUI to manage my server. Any Suggestions? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/96bf8fea/attachment.htm

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

>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)

Hi, I have a Liebert Powersure Personal XT 700 and have followed the installation instructions from here http://www.networkupstools.org/doc/2.0.0/INSTALL.html. I'm pretty sure I've got the ups.conf, upsd.conf and ups.users correct. I have setup upsmon.conf as suggested in the installtion guide with the exception of FINALDELAY which I set to 0 when I found things weren't working

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike