• Stuttering Audio on VOIP phones when first answering a call slow and fast audio


    In this case, Iam assuming you have verified the firewall on either side, and the network conditions are not an issue (eg. no packet loss, no crazy jitter and normal latency). In some phones such as Grandstream or on your PBX, make sure you do NOT use the "adaptive jitter buffer" or similar settings. If possible disable that or change it to "fixed". If you do, you may wonder why when you first start a call the user cannot hear you or the voice s........
  • FreePBX 17 How To Add a Trunk


    ........
  • FreePBX 17 How To Add Phones / Extensions and Register


    Step 1 - Login to FreePBX and Go to Connectivity Click Connectivity -> Extensions Step 2 - Create Extension Click on "Quick Create Extension" or "Add Extension". Quick is fine for more users, unless you have more specific requirements.........
  • How to Install FreePBX 17 in Linux Debian Ubuntu Mint Guide


    FreePBX official install guide is here. Requirements: Debian 12 Download Link Minimal - System Utilities RAM: 4G HDD:20G Note that if you don't have the required base OS yo........
  • 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.........
  • 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........
  • 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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