Quick overview: Register vs Peer Registration (client registers to provider) Peer / No registration Before you start — redaction note All examples below use placeholders: 1) Example — Peer (no registration) trunk (redacted) VICIdial Carrier (Account Entry) — set Protocol = SIP Globals string (in VICIdial globals field) Dialplan example (redacted) Notes 2) Example — Registered trunk (redacted) Registration string (VICIdial Carrier -> Registration String field) (Replace with provider-supplied register => formatted string.) VICIdial Carrier (Account Entry) Globals string (in VICIdial globals field) Dialplan (redacted) Notes Step-by-step in VICIdial GUI (summary) Common troubleshooting tips Security checklist (do these before deploying)…
-
-
🔍 Problem Description While configuring or running VICIdial 11, you may encounter the following error in your Asterisk logs: ⚠️ Cause of the Error This error occurs because the AGI directory (/usr/share/asterisk/agi-bin/) or the AGI scripts (like agi-DID_route.agi) are missing.These AGI files are essential for inbound call routing in VICIdial. 🧩 Solution Steps Follow these commands to fix the issue: After performing these steps, restart Asterisk: Now, VICIdial should be able to execute agi-DID_route.agi without errors, and your inbound calls will route correctly. ✅ Final Check You can verify that the files are properly placed by running: You should see…
-
If you’ve taken a VICIdial backup from one server and restored it on another machine with a different IP address, you’ll need to update the server IP inside VICIdial’s configuration and database tables. Without doing this, agents, phones, and campaigns won’t function correctly. In this post, I’ll explain how I updated my VICIdial IP after moving from 192.168.0.10 to 172.16.11.10. 🔧 Scenario After restoring the database, the system was still pointing to the old IP. To fix this, we must run the VICIdial IP update script. 🪄 Step-by-Step Process 1️⃣ Run the Update Script Open a terminal on your new…
-
How to Connect Your Telco SIP/PRI to a Cloud PBX (Azure/AWS/Datacenter) In many countries, telecom operators provide SIP trunks as a PSTN replacement service, typically delivered over fiber or wireless links. These SIP trunks are usually bound to a private subnet at customer premises, meaning you cannot directly register them from a Cloud PBX in Azure, AWS, or other datacenters. So, how can you securely connect your cloud PBX to your provider’s SIP trunk? Let’s explore two practical options. ⚠️ Disclaimer This post is for informational purposes only. Telecom regulations and licensing differ by country and provider. Always: The Problem…
-
Issabel PBX, a popular open-source Unified Communication solution, saves call recordings in the directory /var/spool/asterisk/monitor/. Over time, these call recordings can consume significant disk space, leading to storage issues. To ensure your server continues to operate efficiently, it is necessary to manage and delete old or unnecessary call recordings periodically. This article explains how to locate and delete Issabel call recording files safely. 1. Understanding the Default Location By default, Issabel PBX stores all call recordings in the following directory: /var/spool/asterisk/monitor/ Within this folder, call recordings are often saved with filenames containing the call date, time, and extension number. For…
-
Interactive Voice Response (IVR) systems are essential for businesses looking to automate customer interactions. A professional and clear IVR prompt ensures that customers can navigate options effectively. If you already have a WAV file for your IVR prompts, you might need to convert it into a specific format compatible with your VoIP system or if you dont have wav files you can create using website Text to WAV converter https://www.narakeet.com/create/text-to-wav.html . In this article, we’ll guide you through creating and converting IVR prompt audio files using two online tools: g711.org and 3CX Audio Converter. Step 1: Understanding IVR Prompt Requirements…
-
How to change SIP Port in Issabel Securing Your Issabel IP PBX: A Step-by-Step Guide to Changing SIP Ports Issabel IP PBX systems play a pivotal role in modern business communications, facilitating seamless voice and data integration. To enhance the security of your Issabel IP PBX, one effective measure is to change the default SIP (Session Initiation Protocol) ports. In this guide, we will walk you through the process of changing the SIP port on your Issabel IP PBX for improved protection against potential security threats. 1- Login to Issabel Web interface go to PBX > PBX Configuration > Setting…
-
How to Change Issabel PBX web interface for Enhanced Security In today’s digital landscape, ensuring the security of your Issabel PBX is crucial. One effective way to enhance security is by changing the default HTTP and HTTPS ports. This step-by-step guide will walk you through the process, making your Issabel PBX less susceptible to potential threats. Step 1: Changing the HTTP Default Port a. Navigate to the HTTP configuration file: Open the file located at /etc/httpd/conf/httpd.conf using your preferred text editor. b. Modify the default HTTP port: Locate the line that reads “Listen 80” and change the port number to…
-
How to configure BSNL SIP Trunk on Asterisk or Issabel BSNL now provides SIP Trunk with DIDs in genreal they call it SIP PRI and asking you to buy channels as they were selling copper PRI links .Now they are providing same service with fiber optice cables on FTTH connections. They are giving fiber link as same they are providing FTTH internet services with ONT and providing PRI service through SIP Protocal over the VPN using OPEN VPN or SoftEther . Depend on your convinent they are asking which VPN service you will use . Here I am using OPENVPN.…
-
Effortless Communication: Connecting Grandstream UCM via SIP Trunk for Seamless Internal Extensions Calling In the realm of unified communication, Grandstream’s UCM series stands out as a reliable and feature-rich solution for businesses seeking seamless connectivity. This article aims to guide you through the process of connecting your Grandstream UCM via SIP trunk, enabling effortless internal extensions calling. Follow these step-by-step instructions to enhance your communication infrastructure and streamline internal collaboration. Below are the youtube link with detailed stepby step explanation. Step 1 – Create SIP Trunk on UCM Site A and UCM and Site B Step 2 – Create Outbound…