Transcribe Asterisk and FreePBX Call Recordings on Your Own Server (faster-whisper, Tested)
A Python script that transcribes Asterisk, FreePBX and Vicidial recordings (.gsm, WAV49, µ-law) on your own CPU server with faster-whisper, splits stereo calls into caller and agent, and writes TXT, SRT and JSON.
October 6, 2026
PJSIP Behind NAT: Asterisk and FreePBX Settings for Two-Way Audio
Configure Asterisk PJSIP and FreePBX behind NAT: local_net and external addresses on the transport, rtp_symmetric, force_rport and rewrite_contact on endpoints, router port forwards and SIP ALG, and how to confirm it with the PJSIP logger.
October 3, 2026
fail2ban for Asterisk and FreePBX: Block SIP Password Guessing
Set up fail2ban for Asterisk PJSIP: turn on the security log, confirm the stock filter matches with fail2ban-regex, create a jail that reads the right files, test a ban, and keep your SIP provider and office off the ban list.
October 3, 2026
How Many SIP Channels Do You Need? Size SIP Trunks with Erlang B
Size SIP trunks properly: measure busy-hour traffic from your PBX CDR, convert it to Erlangs, choose channels with Erlang B for a 1% grade of service, then check bandwidth and set the channel limit on the PBX.
October 3, 2026
Troubleshoot SIP Calls with sngrep and the Asterisk PJSIP Logger
How to use sngrep and the Asterisk PJSIP logger to capture a failing call, read the SIP ladder, save a pcap, check RTP audio, and find NAT, codec and authentication problems on Asterisk and FreePBX.
October 3, 2026
SIP Error Codes on Asterisk and FreePBX: Fix 403, 404, 408, 488 and 503
A troubleshooting walk-through for the SIP errors that break calls on Asterisk, FreePBX and SIP trunks: how to find the failing response, then what causes 403, 404, 408, 480, 488 and 503 and how to fix each.
October 3, 2026
Convert Asterisk Call Recordings to MP3 (FreePBX, Issabel)
How to convert Asterisk, FreePBX and Issabel call recordings from WAV, WAV49 and GSM to MP3: one file with ffmpeg or sox, the whole archive with a script, and every new call automatically, without breaking the call log player.
October 3, 2026
Install Issabel 5 on Rocky Linux 8: Simple Step-by-Step PBX Setup
Prepare a minimal Rocky Linux 8 server, run the Issabel 5 netinstall script, log in to the web interface, and set up extensions, a trunk, routes, the firewall and backups.
September 30, 2026
Install Asterisk 22 LTS on Debian 13: Step-by-Step Guide
Build Asterisk 22 LTS from source on Debian 13, run it as a non-root service, register two PJSIP phones, add a simple dialplan and lock down the SIP and RTP ports.
September 30, 2026
Port Phone Number to VoIP: 6 Steps and Common Rejections Explained
The six steps to move business phone numbers to a VoIP provider or cloud phone system, the documents carriers ask for, the reasons ports get rejected, and how to avoid missed calls on port day.
September 30, 2026
Migrate PBX to Cloud: 8-Step Plan From 3CX or Asterisk
An eight-step plan for moving from an on-premises 3CX, FreePBX or Asterisk system to a hosted phone system: what to inventory, how to test the network, how to pilot and port numbers, and what to keep when you switch the old PBX off.
September 30, 2026
SIP SRV Records: DNS Setup for 3 Transport Types
How SIP clients find a server through DNS (NAPTR, then SRV, then A/AAAA), the exact records to create for UDP, TCP and TLS, how priority and weight give failover, and how to test the result.
September 30, 2026