About AsteriskRecipes
What this site is
AsteriskRecipes is a collection of Asterisk configs, dialplan snippets, shell scripts, and reference docs that you can actually copy and use. Everything here exists because I needed it on a real system at some point. Ring groups, IVRs, PJSIP trunks, one-way audio fixes. That kind of thing.
No fifty-paragraph blog preamble. Just the config, what it does, and enough context that you're not blindly pasting things into your production box.
Why I built it
I've been working with Asterisk for over fifteen years. In that time, I've configured everything from single-office PBX boxes to multi-tenant platforms handling thousands of concurrent calls. And for most of those years, I kept running into the same frustration: the knowledge is out there, but it's scattered across mailing list archives, wiki pages of varying vintage, forum threads that may or may not apply to your version, and blog posts that disappeared when someone forgot to renew their domain.
I started keeping my own notes (working configs, annotated snippets, command references) and eventually realized other people could use them too. AsteriskRecipes is those notes, cleaned up and organized.
Who it's for
Anyone who touches Asterisk. VoIP admins keeping production boxes alive. Engineers who got voluntold into the phone system. Developers writing AGI or ARI apps. Hobbyists running Asterisk at home because they thought "how hard can it be?" (We've all been there.)
If you're dealing with Asterisk, something here will probably save you an hour.
About me
I'm a VoIP engineer. Day to day that means dialplan work, PJSIP configs, carrier interconnects, and making sure nobody's stealing international minutes. The troubleshooting usually starts with a packet capture at 2 AM and ends with a one-line fix you can't believe you didn't see sooner.
Everything on this site has been run on a real system. I don't write entries from the Asterisk wiki alone. If it's here, I've used it or tested it myself.
If you need hands-on help, I do consulting too.
What's coming
New entries show up regularly. Beyond that, I'm expanding the
single-line dialplan builder already on content pages into a full
multi-file config generator with live asterisklint
validation, and eventually sandboxed Asterisk instances you can spin up
in your browser to test things without touching production. Ambitious,
but nobody else is building them.
Recent additions: audio converter improvements (silence trimming, volume normalization, G.722 output), a full reference docs rebuild sourced directly from the Asterisk XML documentation, and a round of security hardening across submissions and the admin interface.
Contributing
Got a config that took you way too long to figure out? Submit it. I review everything before it goes up, but some of the best entries have come from people solving problems I haven't run into yet.
