DFS Replication Backlog: Check It with PowerShell and Fix It
Check the DFS Replication backlog with Get-DfsrBacklog and Get-DfsrState, read the health report, and fix staging quota, sharing violations, event 2213 and 4012.
October 7, 2026
Block RDP Brute Force on Windows Server: Detect and Stop It
Stop RDP brute force on Windows Server: read event 4625 (logon types 3 and 10), close or scope port 3389, require NLA, set account lockout, and auto-block attacking IPs.
October 6, 2026
Open Port Checker: Test TCP Ports on Any Public Server
Check whether TCP ports are open on a public server: up to 10 ports per test, presets for web, mail and control panels, open, closed or filtered results and the SMTP greeting.
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
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 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
VoIP MOS Calculator: Call Quality (R-Factor) from Latency, Jitter and Packet Loss
Estimate the MOS and R-factor of VoIP calls from latency, jitter, packet loss and codec with the ITU-T G.107 E-model, see where the score is lost, and how much loss or delay your network can take.
October 3, 2026
Erlang Calculator: How Many SIP Channels (Erlang B) or Call Centre Agents (Erlang C)
Work out how many SIP channels or trunks you need for a target blocking rate (Erlang B), or how many agents a call centre needs for a service level (Erlang C), from busy-hour calls and average call length.
October 3, 2026
SIP Trace Analyzer: Find NAT, Codec and Dropped-Call Problems in a SIP Log
Paste a SIP trace from sngrep, pjsip set logger, sip set debug or tcpdump and get the call flow, codecs, media addresses and the likely cause of one-way audio, 32-second drops, failed registrations and codec errors.
October 3, 2026
SIP Response Codes: Lookup for Every SIP Error, With Causes and Fixes
Look up any SIP response code (403, 404, 408, 480, 486, 488, 503 and more) or Q.850 hangup cause and see what it means, the usual causes on Asterisk, FreePBX and SIP trunks, and what to check.
October 3, 2026