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
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
JustCall vs CloudTalk: Best Cloud Phone System in 2026 Compared
JustCall and CloudTalk compared on price, minimum licences, calling allowances, number coverage, integrations, dialers and AI. JustCall suits CRM-driven teams of two or more; CloudTalk suits routing-heavy teams, single users and multi-country numbers.
September 30, 2026