• Cisco How To Use a Third Party SIP Phone (eg. Avaya, 3CX)


    Most relevant config points from my video here. 1.) Create a new End User The most important part is setting the "Digest Credentials", that is the password that the phone will use to authenticate.........
  • How to convert audio for Asterisk .wav format


    In our example we take "sound.mp3" and convert it to .wav. Generally Asterisk for its wave needs one audio channel (-ac 1) / mono and 8000hz (-ar 8000) instead of the standard CD/MP3 of 44100hz. Here is the command to convert into Asterisk .wav format: ffmpeg -i sound.mp3 -ac 1 -ar 8000 sound.wav Errors Asterisk may give you if the format is wrong: -- Executing [91781891@cme:3] Playback("SIP/234-000........
  • Using Cisco CME Router with Asterisk as a dial-peer


    #Remember that you need a valid gateway IPunless the Asterisk server is on the same subnet and LAN Set Valid Gateway IP (if you don't have one already) ip route 0.0.0.0 0.0.0.0 GATEWAYIP Enable VOIPTrust voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 sip Set Credentials For Asterisk and Register To Asterisk sip-ua........
  • Cisco CME How To Configure SIP Trunk VOIP


    This is how we configure outside PSTNaccess or dialing through another SIPtrunk. Beware that this a simplistic example that neglects most security including SRTP First you'll need to be in config mode: Step - 1 Enter Voice Service VOIP security options There are more options but for now we'll just focus on security/allowing connections to and from our phones and the trunk. Router(config)#voice service voip ........
  • apcupsd how to setup and monitor APC UPS units


    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........
  • Asterisk Does Not Retry When Authentication Fails


    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&........
  • How To Install Asterisk 16 17 on Debian Ubuntu Linux


    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........
  • How Does Cisco CUCM (Cisco Unified Communication Manager) Work?


    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.........
  • Linux Mint Black Screen after boot no graphics solution


    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........
  • ffmpeg Linux Mint download, compile and install howto


    #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 ........
  • How to clear all iptables rules for all tables and chains


    iptables -F iptables -t nat -F iptables -t mangle -F This is as simple as it gets.........
  • Sipura/Linksys VOIP Adapter How to Check your IP Address


    Dial **** Then 110# It will then say what your current IP address is.........
  • Asterisk VOIP Sipura/Linksys PAP2T Calls Not Being Received Solution


    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)? solution


    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 Adventures, Guide and HowTo


    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........
  • Asterisk DOS attack - failed for '173.242.117.192' - Peer is not supposed to register [May 23 15:46:07] ERROR[32748]: chan_sip.c:13158 register_verify: Peer '153' is trying to register, but not configured as host=dynamic


    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.........
  • Asterisk - Peer is not supposed to register


    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........
  • VOIP/Asterisk/FAX Error Problems Solution NOTICE[11389]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 100 received from


    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 Horrible


    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........
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