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
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
Asterisk Recordings to MP3: Bulk Convert FreePBX Call Recordings
Converts Asterisk, FreePBX and Issabel call recordings (WAV, WAV49, GSM, µ-law, SLN) to small mono MP3 files from cron, verifies each file and keeps the FreePBX call log player working.
October 3, 2026
Call Recording Converter: GSM, WAV49, µ-law and SLN to MP3
Converts Asterisk, FreePBX, Issabel and Vicidial call recordings (.gsm, GSM-in-WAV, µ-law, A-law, .sln, .au) to MP3 or WAV in batches, in your browser, with nothing uploaded.
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
VoIP Bandwidth Calculator: Per-Call and Total Bandwidth by Codec
Works out how much upload and download a number of concurrent calls needs for a given codec, packet interval, IP version, SRTP and VPN tunnel, with a byte-by-byte breakdown.
September 30, 2026
SIP SRV Checker: Free NAPTR, SRV and SIP TLS Test
Checks the DNS records SIP phones and trunks use to find your server: NAPTR transport preference, SRV records for UDP, TCP and TLS, and whether every target resolves correctly.
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