It really seems limited in that it can mainly give you the things you would see on the physical unit such as load etc.. wget https://downloads.sourceforge.net/project/apcupsd/apcupsd%20-%20Stable/3.14.14/apcupsd-3.14.14.tar.gz?r=https%3A%2F%2Fsourceforge.net%2Fprojects%2Fapcupsd%2Ffiles%2Flatest%2Fdownload&ts=1598115866 tar -zxvf apcupsd-3.14.14.tar.gz cd apcupsd-3.14.14 [root@somebox apcupsd-3.14.14]# ./conf........

When authentication times out that is one thing, but when it just fails like below Asterisk by default will not re-register until you the admin reload the sip or asterisk server: voipserver*CLI> sip show registry Host dnsmgr Username Refresh State&........

Downloading and compiling from source to get the latest version of Asterisk is really simple with this guide. apt install gcc make g++ libedit-dev uuid-dev libjansson-dev apt install libxml2-dev sqlite3 libsqlite3-dev wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16-current.tar.gz tar -zxvf asterisk-16-current.tar.gz cd asterisk-16.6.2/ ./configure If you get this error change y........

Cisco's CUCM (Cisco Unified Communication Manager) is a system that combines voice, video, data and mobile products into a single unified management suite. At its core, the CUCMis like a "Super PBX" that controls the flow of all communications through an organization even single or multiple site deployments. Cisco's CUCMmakes communication more effective and simple through centralized management and unification of communications resources.........

This is not the normal "black screen"issue and I was shocked to eventually find out why. The normal advice of reconfiguring Xorg didn't work. Even booting into "Recovery Mode" did not help. Here is the short end of the stick that fixed it: sudo apt-get install mdm mate-desktop-environment Yes you got it right, mdm and the mate-desktop-environment / gnome were somehow uninstalled. This must be whe........

#if you have nvidia make sure you install the nvidia-cuda-toolkit so hardware acceleration can be used wget http://ffmpeg.org/releases/ffmpeg-3.3.2.tar.bz2 tar -jxvf ffmpeg-3.3.2.tar.bz2 cd ffmpeg-3.3.2/ ./configure --disable-yasm install prefix /usr/local source path ........

iptables -F iptables -t nat -F iptables -t mangle -F This is as simple as it gets.........

Dial **** Then 110# It will then say what your current IP address is.........

This problem seemed to happen recently but was likely causing issues before where the phone(s) do not ring. Now there are a few reasons why this can happen especially if your adapter has DND mode enabled (disable it). However that wasn't my issue and Ionly figured it out the other day when by fluke if you're on the phone (making a call) then calls will come in. That's when Ifigured out the solution: This likely app........

Sipura / Linksys PAP/VOIP/SIP Adapter Issue Can't receiving incoming phone calls and you're behind a NAT router (99% of people)? 1. Login to the adapter. 2. Click on "Advanced" (location varies but usually somewhere on top) 2. Click on "SIP" 3. Scroll down to "NAT" (usually at the bottom). You'll find 2 columns with 4 rows of drop-down boxes (they'........

LSi Megaraid At first it was configured as a RAID 0, then I deleted the Virtual Disk Group. I thought both drives would be shown and detected in Linux as sda and sdb but it actually shows nothing. To make them work you have to hit Ctrl+R before the system boots (when prompted) and create a Virtual Disk Group. In my case I created each one as RAID 0 (with a single drive only) as I just wanted JBOD but there is no such option or default in these Dell Pe........

This happened to a customer Asterisk server and it somehow found the ID of the registration account to the upstream SIP server and was railing connection attempts (it filled up the console and there were literally thousands per second). Basically this caused all incoming and outgoing calls to fail. It was a temporary fix but the solution was to block that specific IP, it's hard to stop it 100% because the customer needs the default SIP port.........

NOTICE[5628] chan_sip.c: Registration from 'user ' failed for '192.168.5.25' - Peer is not supposed to register You have to setup as a "friend" and not "peer". Set: type=friend........

NOTICE[11389]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 100 received from I've found that trying several times may work, but I also read changing your Sipura VOIP adapter settings as follows helps (but it still fails for me sometimes): In adapter change the following under SIP/Advanced Codec to G711a from G711u Passthru Method from: NSE to ReINVITE FAX CED Detect Enable: Yes FAX CNG Detect........

Telus + 2Wire 2700 Router HorribleWell first of all let me say this is the only router/switch that sometimes seems to crash/disconnect computers on the local network. This device also thought it would be smart to block VOIP packets coming from my Sipura ATA VOIP adapters so I disabled the [quote:cb89ba7bff]"Invalid TCP Flag Attacks (NULL/XMAS/Other)"[/quote:cb89ba7bff] option Then all of a sudden I couldn't get onto any web pages, the wireless........