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VoIP and Telephony Guides, Tools and Scripts
Asterisk, FreePBX, Vicidial and FusionPBX: install guides, SIP and PJSIP fixes, free browser tools and call-recording scripts in one place.
Short answer: This is the srvScripts hub for VoIP and telephony admins: tested guides for Asterisk, FreePBX, Vicidial and FusionPBX, free browser tools for SIP and PJSIP problems, and scripts for call recordings. Start with the section that matches your problem; every guide lists the exact commands and the versions we checked them on.
Most guides were tested on our FreePBX 17 / Asterisk 22 lab server; each page says exactly what was run and what was checked against the official documentation.
Table of Contents
Install and upgrade
- Install FreePBX 17 on Debian 12 (Open-Source Only, Tested)
- Install FusionPBX on Debian 12 with the Official Installer
- Install VICIdial with ViciBox 12: Express Setup, SSL and Firewall
- Asterisk Versions and EOL Dates: LTS Support Table and Upgrade Plan
- Upgrade FreePBX 16 to 17: Backup, Restore and Migration Checklist
- Install Issabel 5 on Rocky Linux 8: Simple Step-by-Step PBX Setup
- Install Asterisk 22 LTS on Debian 13: Step-by-Step Guide
- Port Phone Number to VoIP: 6 Steps and Common Rejections Explained
- Migrate PBX to Cloud: 8-Step Plan From 3CX or Asterisk
Audio, NAT and SIP problems
- VICIdial “No one is in your session” Error: 8 Checks to Fix It
- Asterisk WebRTC Setup: WSS Transport, Certificates, webrtc=yes
- PJSIP Endpoint Unreachable or Unavailable: Fix Qualify and NAT
- PJSIP Behind NAT: Asterisk and FreePBX Settings for Two-Way Audio
- Troubleshoot SIP Calls with sngrep and the Asterisk PJSIP Logger
- SIP Error Codes on Asterisk and FreePBX: Fix 403, 404, 408, 488 and 503
- SIP SRV Records: DNS Setup for 3 Transport Types
- VoIP One-Way Audio Fix: 6 Checks for NAT and RTP
- Disable SIP ALG on 9 Routers and Firewalls (One-Way Audio Fix)
- SIP Ports Firewall Rules: 4 Setups for CSF, firewalld and pfSense
Security and fraud
- STIR/SHAKEN for Asterisk and FreePBX: Attestation and Who Signs
- Asterisk and FreePBX Toll Fraud Prevention: 10-Point Checklist
- FreePBX Responsive Firewall vs Intrusion Detection (FreePBX 17)
- fail2ban for Asterisk and FreePBX: Block SIP Password Guessing
Trunks, dialplan and configuration
- Asterisk AudioSocket: Connect a Call to an AI Voice Agent
- Asterisk Dialplan Pattern Matching: X, N, Z, [ ], . and ! Explained
- FreePBX PJSIP Trunk Setup: Registration, IP Auth, Routes, NAT
- Asterisk CLI Commands Cheat Sheet: PJSIP, Calls, Dialplan, Logs
- chan_sip to PJSIP Migration: sip_to_pjsip.py, Option Mapping, Tests
Call recording and transcription
- Transcribe Asterisk and FreePBX Call Recordings on Your Own Server (faster-whisper, Tested)
- Asterisk Call Recording with MixMonitor: Formats, Stereo, Retention
- Convert Asterisk Call Recordings to MP3 (FreePBX, Issabel)
Capacity and call quality
- How Many SIP Channels Do You Need? Size SIP Trunks with Erlang B
- VoIP Call Quality: Jitter, Packet Loss and MOS Explained With 3 Targets
- VoIP Codec Bandwidth: G.711 vs G.729 vs Opus, Per-Call Figures
Comparisons
Free VoIP tools
- STIR/SHAKEN Identity Header Decoder (PASSporT)
- Asterisk Dialplan Pattern Tester: Which Extension Matches?
- sip.conf to pjsip.conf Converter for Asterisk
- Call Recording Transcription for Asterisk, FreePBX and Vicidial
- PJSIP NAT Settings Generator: Asterisk, FreePBX, Issabel, VitalPBX, 3CX
- SIP Firewall Rules Generator: iptables, nftables, UFW, firewalld, CSF and pfSense
- VoIP MOS Calculator: Call Quality (R-Factor) from Latency, Jitter and Packet Loss
- Erlang Calculator: How Many SIP Channels (Erlang B) or Call Centre Agents (Erlang C)
- SIP Trace Analyzer: Find NAT, Codec and Dropped-Call Problems in a SIP Log
- SIP Response Codes: Lookup for Every SIP Error, With Causes and Fixes
- Call Recording Converter: GSM, WAV49, µ-law and SLN to MP3
- VoIP Bandwidth Calculator: Per-Call and Total Bandwidth by Codec
- SIP SRV Checker: Free NAPTR, SRV and SIP TLS Test