VICIdial “No one is in your session” Error: 8 Checks to Fix It
Fix the VICIdial "No one is in your session" error: screen processes, phone registration and server IP, conference entries, DAHDI timing, ViciPhone SSL and WebSocket.
October 6, 2026
Install FusionPBX on Debian 12 with the Official Installer
Install FusionPBX on Debian 12 with the official script: review config.sh, what the installer changes, saving the admin login, first hardening steps and checks.
October 6, 2026
Install FreePBX 17 on Debian 12 (Open-Source Only, Tested)
Install FreePBX 17 with Asterisk 22 on Debian 12 using the official script, get past the oracle_connector failure with --opensourceonly, secure the admin login and fix fail2ban. Tested 6 Oct 2026.
October 6, 2026
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
Call Recording Transcription for Asterisk, FreePBX and Vicidial
Upload an Asterisk, FreePBX or Vicidial call recording (.gsm, WAV49, .wav, µ-law, A-law) and get a timestamped transcript with caller and agent separated. TXT, SRT and JSON downloads.
October 6, 2026
PJSIP NAT Settings Generator: Asterisk, FreePBX, Issabel, VitalPBX, 3CX
Generate the PJSIP NAT settings for Asterisk or FreePBX: transport local_net and external addresses, endpoint rtp_symmetric, force_rport and rewrite_contact, rtp.conf, router port forwards and the commands that prove it works.
October 3, 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
SIP Firewall Rules Generator: iptables, nftables, UFW, firewalld, CSF and pfSense
Generate firewall rules for Asterisk, FreePBX and other SIP servers: SIP limited to your provider and phones, the RTP range open, scanner protection, and the matching rtp.conf and PJSIP NAT settings.
October 3, 2026