Converting Audio Files for Asterisk: Formats, sox and Sample Rates
Why your prompt plays as noise or not at all, the formats Asterisk expects, and the exact sox commands to convert audio for prompts, IVR and music on hold.
Himanshu Pal
Asterisk Originate: Click-to-Call from the CLI, AMI and Call Files
How to make Asterisk place a call programmatically — the CLI syntax, every AMI Originate header that matters, and why your originated calls time out after 30 seconds.
Asterisk CDR with MySQL: Storing and Querying Call Records Properly
What each CDR field actually contains, which ones you can modify, how to store records in MySQL, and the queries that answer real billing and reporting questions.
Asterisk Call Queues: Ring Strategies, Members and Realistic Configuration
How Asterisk queues actually distribute calls — every ring strategy explained, the settings that matter in queues.conf, and the mistakes that make queues behave unexpectedly.
Jitter Buffers Explained: Why VoIP Audio Stutters and How to Tune It
What jitter actually is, how a jitter buffer trades latency for smooth audio, the difference between fixed and adaptive buffers, and how to configure one in Asterisk.
MOS Score Explained: How VoIP Call Quality Is Actually Measured
What Mean Opinion Score means, where the 1-to-5 scale comes from, what counts as an acceptable score, and how MOS is estimated automatically on live calls.
QoS for VoIP: DSCP Marking, Priority Queues and What Actually Helps
How to mark and prioritise VoIP traffic properly — the DSCP values the standards recommend, where marking works, and why QoS cannot fix an internet connection.
VoIP Codecs Compared: G.711, G.729, G.722 and Opus
A practical comparison of the codecs you will actually meet in VoIP — bandwidth, quality, licensing and when each is the right default.
SIP REFER and Call Transfers: Blind vs Attended, and Why They Fail
How SIP REFER actually performs a call transfer, the difference between blind and attended transfers, and the usual reasons a transfer drops the call instead of completing it.
RTP Ports and Firewalls: The Real Reason Your VoIP Calls Have No Audio
Signalling and media use completely different ports in SIP. Here is how RTP port ranges work, how to configure them in Asterisk, and how to open them correctly on your firewall.
